Commit graph

4700 commits

Author SHA1 Message Date
Sebastian Dröge
641428966e audiorate: Use gst_audio_format_fill_silence() instead of memset with 0 for generating silence
For unsigned formats, silence is not all bits 0.
2016-01-28 13:29:39 +01:00
Thibault Saunier
135c612550 encodebin: Allow streamheader update when profile.allow_dynamic_output == FALSE
Some encoders can update the stream header through time (for example
vp8 might do that) but it does not strictly changes the output format.
2016-01-27 12:58:23 +01:00
Sebastian Dröge
acd08a828d decodebin: Correctly expose pads from elements that have directly exposable pads
analyze_new_pad() can return a new decode chain, which might have a new
GstDecodePad in the end. We should use those two for expose_pad() and not the
original ones that were passed to analyze_new_pad().

This fails when having a demuxer element that has raw pads immediately or
if a decoder with raw caps is after an adaptive demuxer.

https://bugzilla.gnome.org/show_bug.cgi?id=760949
2016-01-25 13:50:26 +01:00
Sebastian Dröge
a7b86878fb audio: Move audioaggregator base class to a library
It's useful enough already to be used in other elements for audio aggregation,
let's give people the opportunity to use it and give it some API testing.

https://bugzilla.gnome.org/show_bug.cgi?id=760733
2016-01-22 12:39:48 +02:00
Mathieu Duponchelle
2717f4a86f streamsynchronizer: Ignore flushing streams [..]
[..] when resetting group start time. In GES, we are usually connected
to the streamsynchronizer on one audio and one video pad.

When seeking the timeline, both nlecompositions often output their flush_start
before any of them has output its flush_stop.

The current code, when receiving the first flush stop was using the
running time of the start of the second composition, which could
be pretty much anything, and means nothing at that point.

This patch is thread-safe, as STREAM_SYNCHRONIZER_LOCK is taken
both when setting flushing and when checking it.

https://bugzilla.gnome.org/show_bug.cgi?id=750013
2016-01-16 11:05:13 +01:00
Sebastian Dröge
fccf83e69f playbin: Only append non-raw and sysmem pad template caps to the autoplug-query result
Otherwise a decoder supporting GL memory will think that all downstream can
support GL memory because of seeing its own template caps.

https://bugzilla.gnome.org/show_bug.cgi?id=758212
2016-01-16 11:05:13 +01:00
Sebastian Dröge
9713ab06cd Revert "playbin: only add the template caps when the result is empty"
This reverts commit 023af2d3b1.

https://bugzilla.gnome.org/show_bug.cgi?id=758212
2016-01-16 11:05:13 +01:00
Edward Hervey
62053852de playsink: Properly mark pending blocked pads
When blocking input pads, we also need to properly set the appropriate
pending flag.

Without this, when switching stream types after initial configuration
(like going from Audio+Video to Audio+Video+Sub) playsink would never
wait for *all* input streams to be blocked (it would just wait for the
new input pad (text in this case) to be blocked).

Since the reconfiguration might introduce unlinking/relinking of elements,
we need to ensure that *ALL* input streams are blocked.

Failure to do so would result in having some input streams pushing data
to inactive elements (returning GST_FLOW_FLUSHING) or unlinked pads
(returning GST_FLOW_NOT_LINKED).

A later optimization could involve only blocking the input pads that
might be involved in reconfiguration. But better be safe than sorry for
now :)
2016-01-15 10:05:58 +01:00
Thiago Santos
0d18717912 subtitleoverlay: replace gst_caps_can_intersect() with is_subset()
Subset check verifies also that all required fields are present
and is mostly commonly used when checking if an element accepts
a certain caps
2016-01-13 16:32:25 -03:00
Thiago Santos
81c52aaa16 playbin: use subset check instead of intersect
Elements usually require that all fields on their caps are present
on the fixed caps they receive. Using intersection won't verify it,
resort to using is_subset() checks.

https://bugzilla.gnome.org/show_bug.cgi?id=760477
2016-01-13 15:29:17 -03:00
Thiago Santos
20f6af651b subtitleoverlay: replace accept-caps with caps query
Those accept caps are actually checking if downstream supports
some particular caps to check if it need to negotiate a different
format. Checking only the next element with accept-caps is not enough
to guarantee that it is supported.

Using a caps query makes it obtain the supported caps for downstream
as a whole instead of only the next element.
2016-01-11 18:35:29 -03:00
Thiago Santos
5ef0a09794 videorate: replace accept-caps with a caps query
accept-caps is only a shallow check, it needs to know
whether downstream as a whole accepts the framerate
2016-01-08 15:05:38 -03:00
Wim Taymans
85afad72ec audio-converter: small API tweaks
Pass flags in _converter_new() so that we can configure ourselves
differently depending on some options.
SOURCE_WRITABLE -> IN_WRITABLE because the array is called 'in'
2016-01-08 17:34:50 +01:00
Wim Taymans
980163457e audio-convert: simplify API
Simplify the API, we don't need the consumed and produced output
arguments. The caller needs to use the _get_in_frames/get_out_frames API
to check how much input is needed and how much output will be produced.
2016-01-08 17:19:58 +01:00
Sebastian Dröge
844aa3e6a9 playbin: Use the caps query instead of accept-caps to detect if a sink accepts caps
accept-caps is only for one element, caps query is recursive. Fixes playback
with totem and other situations.

https://bugzilla.gnome.org/show_bug.cgi?id=760234
2016-01-08 16:32:32 +02:00
Aurélien Zanelli
9b9f913809 videotestsrc: add missing break in set_property switch case
To avoid future issue when adding new properties.

https://bugzilla.gnome.org/show_bug.cgi?id=760204
2016-01-06 13:21:06 +02:00
Sebastian Dröge
eb09889176 audioconvert: Pass pointer arrays instead of singleton pointers to gst_audio_converter_samples()
In this specific case it wouldn't cause problems as we only ever access the
first array element, but let's make explicit what is happening here.

CID 1346530 and 1346529
2015-12-29 18:14:54 +02:00
Sebastian Dröge
0416f121f2 typefindfunctions: Make sure that enough data is available in AAC/ADTS typefinder
We would otherwise read beyond the array bounds and crash every now and then.
This was introduced with 5640ba17c8.

https://bugzilla.gnome.org/show_bug.cgi?id=759910
2015-12-28 13:51:02 +02:00
Sebastian Dröge
6a57399270 playsink: Don't leak audio/video filters due to floating references weirdness
The filters' floating references are sinked during set_property() already,
which means that GstBin takes a new reference when adding the filter to it.
Get rid of the additional reference after adding the filter to the bin.
2015-12-25 11:34:10 +01:00
Sebastian Dröge
a136ac0e2f playsink: Allow reuse of audio/video filters by unparenting them from their bins
And also recreate the chains if the filter is changing.
2015-12-25 10:36:44 +01:00
Sebastian Dröge
24181db083 playsink: Don't leak audio/video filters when using non-raw media 2015-12-25 10:28:02 +01:00
Matthew Waters
023af2d3b1 playbin: only add the template caps when the result is empty
Unconditionally adding the template caps when proxying the caps query will play
havoc with decoders that attempt to choose an output format based on some caps
features.  Creating a sink that does not include those caps features and a
decoder/parser/etc that preferentially chooses some specific caps feature when
available, will always return the decoder/parser/etc template caps and choose a
feature that downstream will be unable to support.

Fix by limiting the addition of the template caps to when the result is actually
empty.

https://bugzilla.gnome.org/show_bug.cgi?id=758212
2015-12-18 21:55:00 +11:00
Sebastian Dröge
60bad4815d Revert "decodebin2: fix deadlock on chain shutdown"
This reverts commit 77dc09c3a9.

It can cause the FLUSH_START/STOP events to go to the sink elements, which
then causes state changes and various other problems. We shouldn't really
flush downstream here, the idea is to do *draining*.

Apart from that the testcase for the original bug here works without this
commit now.
2015-12-16 17:09:25 +01:00
Luis de Bethencourt
29cfb9a6d7 multifdsink: fix typo in GST_WARNING_OBJECT
This should make easier to parse the debug logs.
s/fnctl/fcntl
2015-12-16 11:12:03 +00:00
Vincent Penquerc'h
033ce9b20d videorate: remove dead code
Since the loops increasing count from 0 are always run at least
once (if count < 1), count will always be at least one when
compared to the drop/dup conditions.

Coverity 1139674
2015-12-16 11:00:22 +00:00
Wim Taymans
8bcf183c7f audioconvert: clear convert object 2015-12-16 11:13:15 +01:00
Vineeth TM
f6cd84c4e6 plugins-bad: Fix example pipelines
rename gst-launch --> gst-launch-1.0
replace old elements with new elements(ffmpegcolorspace -> videoconvert, ffenc_** -> avenc_**)
fix caps in examples

https://bugzilla.gnome.org/show_bug.cgi?id=759432
2015-12-15 10:30:49 +00:00
Wim Taymans
f5a3f70571 audio: adapt API for non-interleaved formats
Allow an array of sample blocks to be passed to the channel mix and
quantizer functions to support non-interleaved formats.
2015-12-14 09:16:08 +01:00
Wim Taymans
9c2bcd7b76 multisocketsink: add GstNetworkMessage event
Add a property and logic to send a GstNetworkMessage event containing
the message that was received from a client. This can be used to
implement simply bidirectional communication.
2015-12-10 12:44:42 +01:00
Wim Taymans
9aaaa26ff3 multisocketsink: add dispatched event
Add a property and logic to send a GstNetworkMessageDispatched
event upstream to notify that a buffer has been sent. This can be used
to keep track of what client received what buffers.
2015-12-10 12:44:42 +01:00
Wim Taymans
0e1a858d89 socketsrc: handle GstNetworkMessage events
Add a property to handle GstNetworkMessage events. These events contain
a buffer that is sent on the socket to allow for simple bidirectional
communication.
2015-12-10 12:44:42 +01:00
Wim Taymans
5e55968546 audio-convert: improve converter API
Improve the converter API to allow for an max input and output number of
samples and return the number of consumed/produced samples.
2015-12-09 17:16:26 +01:00
Wim Taymans
1da5a3ab66 multisocketsink: let downstream know we support metadata
Let downstream know that we support GstNetControlMessage metadata API.
2015-12-04 12:25:11 +01:00
Tim-Philipp Müller
71505dfa24 decodebin2: fix "Attempt to unlock mutex that was not locked"
Introduced in commit ee44337f, caused the decodebin
test_text_plain_streams unit test to abort.

https://bugzilla.gnome.org/show_bug.cgi?id=752651
2015-12-02 18:16:05 +00:00
Edward Hervey
d292ed48c5 playback: Expose XSUB formats by default
This is a workaround, we should remove this once we have a proper
decoder
2015-12-02 16:37:50 +01:00
Edward Hervey
c79bf13bc2 streamsynchronizer: Rename GstStream => GstSyncStream
Avoid clashes with future GstStream from core
2015-12-02 16:37:41 +01:00
Sebastian Dröge
9e4bf58b8e decodebin: Update buffering messages when removing an element that had buffering pending
Otherwise we'll remove that element while keeping its buffering message in our
list, and because of that never ever report buffering 100% as that element
will always be at a lower percentage.

This fixes e.g. seeking over Period boundaries in DASH and various other
issues when buffering happens between group switches.

Also use a new mutex for protecting the buffering messages. The object lock is
already used by gst_object_has_as_ancestor() and we need to use it now for
checking if the buffering message sender has the to-be-removed element as
ancestor.
2015-12-02 16:16:22 +02:00
Wim Taymans
01f5ca3da8 multisocketsink: keep on reading when we stop sending
When we stop sending because we need more data, still keep a GSource
around to receive data from the clients.
Also handle read and write in the same go.
2015-12-02 10:26:03 +01:00
Thomas Bluemel
2c62aad159 [PATCH] Fix a race condition accessing the decode_chain field.
Make sure that any access to the GstDecodeBin's decode_chain
field is protected using the EXPOSE_LOCK.  Also add a simple
reference counter to the GstDecodeChain structure so that when
the type_found signal fires it can hold onto the decode chain
even while the EXPOSE_LOCK is not held.  This should fix a
race condition if the type_found signal fires right in the
middle of a state change that messes with the same decode
chain.

https://bugzilla.gnome.org/show_bug.cgi?id=755260
2015-12-01 17:36:31 +00:00
Vincent Penquerc'h
870c6df489 decodebin: early out on pad-added when the pad is inactive
The pad may be recently deactivated if the element is switched
back down very quickly.

https://bugzilla.gnome.org/show_bug.cgi?id=752651
2015-12-01 17:36:31 +00:00
Vincent Penquerc'h
ee44337fc3 decodebin: lock the expose lock around decode_chain use
Helps with a crash in decodebin when quickly switching states.

https://bugzilla.gnome.org/show_bug.cgi?id=752651
2015-12-01 17:36:31 +00:00
Tim-Philipp Müller
9f69e97935 audiomixer: register function name for debugging just once
Not every time aggregate is called...
2015-11-24 15:17:30 +00:00
Wim Taymans
ff6d1a2a25 audio-converter: add output size argument
Make it possible to have a different number of output samples than input
samples when we, for example, want to add resampling later.
2015-11-10 09:53:59 +01:00
Edward Hervey
d0eface01c decodebin: Properly deactivate ghostpads
Just setting the ghostpad as flushing wasn't enough. It needs to be
consistent on the internal proxypad also, otherwise you end up in
situations where:
* a pending buffer on the target pad triggers the sticky event
  propagation
* the default implementation sees that the proxypad is not flushing,
  so it tries to push it to the other pad (the actual ghostpad)
* the ghostpad is flushing, so returns FALSE
* the push_event function sees that pushing the event failed...
* ... and pending buffer push returns GST_FLOW_ERROR, instead of
  GST_FLOW_FLUSHING

By using gst_pad_set_active(FALSE), we ensure that both the ghostpad
and the proxypad are flushing/deactivated. The situation above will
no longer occur, and a GST_FLOW_FLUSHING will be returned.
2015-11-06 19:38:13 +01:00
Tim-Philipp Müller
d2e210bbea audioconvert: fix build
Don't include file that is no longer generated, and remove some
files that are no longer needed because they have moved into the
lib. Fixes distcheck.
2015-11-06 18:12:28 +00:00
Wim Taymans
e3f0f3b91e audio-converter: move audio converter to audio libs
Move the audio-converter helper to the audio library.
2015-11-06 17:53:22 +01:00
Wim Taymans
dfa25a40fc audio-channel-mix: move channel mixer to audio libs
Move the channel mixer code to the audio library
2015-11-06 17:39:33 +01:00
Wim Taymans
b8bea9d8be audio: add debug categories 2015-11-06 17:29:22 +01:00
Wim Taymans
268ed5dd6f channelmix: don't limit channelpositions
Don't set a limit on the channel positions, just like the metadata.
2015-11-06 16:42:35 +01:00
Wim Taymans
9fbe0386d0 channelmix: simplify API a little
Remove the format and layout from the mix_samples function and use the
format when creating the channel mixer object. Also use a flag to handle
the unlikely case of non-interleaved samples like we do elsewhere.
2015-11-06 16:03:20 +01:00
Wim Taymans
7f5104f52f channelmix: GstChannel -> GstAudioChannel
Rename GstChannel to GstAudioChannel
2015-11-06 15:50:34 +01:00
Wim Taymans
1635bc0a45 audioconvert: cleanups and add some docs
Add docs for the internal audioconvert object before moving it to the
audio library.
Remove get_sizes and implement the trivial logic in the element.
Remove some unused orc functions
2015-11-06 12:46:36 +01:00
Wim Taymans
c36ac3ce45 audioconvert: move audio quantize code to libs
Move the audio quantize code from audioconvert to the audio library.
work on making an audio converter helper function similar to the video
converter.
Fold fastrandom directly into the quantizer, add some ORC code to
optimize this later.
2015-11-06 12:10:48 +01:00
Wim Taymans
a7789854d5 audio-channels: rename get_default_mask
Rename _get_default_mask() to _get_fallback_mask() to make it more
clear that the function only provides a fallback if nothing else can be
done. Also clarify this in the documentation.

API: gst_audio_channel_get_fallback_mask()
2015-11-05 12:50:18 +01:00
Thibault Saunier
9c7d3c8ab2 volume: Do not try to get binding value array if we are not processing any sample
In some conditions we might process empty buffers, calling
gst_control_binding_get_value_array in that case will lead
to the assertion:

  (lt-ges-launch-1.0:18859): GStreamer-CRITICAL **: gst_control_binding_get_value_array: assertion 'values' failed
2015-11-05 11:44:31 +01:00
Wim Taymans
f86ed8cdf6 audio-channels: make method to get default channel-mask
Add a new method to get the default channel-mask.
Use the new method on audiodecoder and audioconvert.

API: gst_audio_channel_get_default_mask()
2015-11-05 10:52:53 +01:00
Wim Taymans
801f7ca464 audio-format: add TRUNCATE_RANGE flag
Add a TRUNCATE_RANGE flag for unpack functions to fill the least
significate bits with 0 (as did the old code). Also add functions
that don't truncate. Use the TRUNC flag in audioconvert for
backwards compatibility for now.
2015-11-03 12:12:08 +01:00
Wim Taymans
9e15c89564 audioconvert: change multiplier for int<->float conversion
Use (1 << 31) as the multiplier for int<->float conversions. This makes
sure that int->float conversions always end up with floats between
[-1.0, 1.0].
For the conversion from float to int, this multiplier will give the complete
int range after we perform clipping.
Change the unit test to take this into consideration.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755301
2015-11-03 12:12:08 +01:00
Olivier Crête
1369924fa0 audioaggregator: Improve log messages
Make the level of log messages saner and improve some.
2015-11-02 19:40:28 -05:00
Wim Taymans
bd89f2430b audiotestsrc: increase freq limit
Raise the frequency limit and try to negotiate to a samplerate of 4*freq
when larger then the default samplerate.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=754450
2015-11-02 15:54:19 +01:00
Wim Taymans
c688eb0d88 audiotestsrc: add support for unlimited number of channels
Raise the channel limit and set the channel-mask for > 2 channels.
2015-11-02 15:46:22 +01:00
Wim Taymans
b0bf294a62 audiotestsrc: add support for all formats
Use the pack functions to also support the other audio formats we
have.
2015-11-02 13:22:18 +01:00
Sebastian Dröge
e51c9a3dad audioresample: Clip input buffers to the segment before handling them
https://bugzilla.gnome.org/show_bug.cgi?id=757068
2015-11-02 10:20:37 +02:00
Sebastian Dröge
000c424835 audioconvert: Clip input buffers to the segment before handling them
https://bugzilla.gnome.org/show_bug.cgi?id=757068
2015-11-02 10:20:37 +02:00
Tim-Philipp Müller
3dd26bb9e8 audioconvert: update orc backup code to fix build without orc 2015-11-01 23:06:11 +00:00
Csaba Toth
3159501002 multisocketsink: fix "client-removed" signal on 64-bit platforms and with bindings
The client-removed signal used G_INT_TYPE instead of G_SOCKET_TYPE
in its definition leading to problems on platforms where the size
of a pointer is larger than the size of an integer, It would also
not work at all with dynamic language bindings.

https://bugzilla.gnome.org/show_bug.cgi?id=757155
2015-10-31 11:12:38 +00:00
Joan Pau Beltran
a95a900c21 videotestsrc: fix handling of Bayer format 'gbrg'
Due to a typo, videotestsrc did not handle the Bayer
format 'gbrg' properly and reported it as invalid,
causing negotiation errors.

https://bugzilla.gnome.org/show_bug.cgi?id=757264
2015-10-30 20:29:04 +00:00
Wim Taymans
5cf367ae57 audioconvert: rework audioconvert
Rewrite audioconvert to try to make it more clear what steps are
executed during conversion.
Add passthrough step that just does a memcpy when possible.
Add ORC optimized dither and quantization functions.
Implement noise-shaping on S32 samples only and allow for arbitrary
noise shaping coefficients if we want this later.
2015-10-30 17:51:47 +01:00
Wim Taymans
e1569ce76a channelmix: fix up API a little
don't use gpointer * for something that should be gpointer.
2015-10-30 17:51:47 +01:00
Wim Taymans
26d469a04b audioquantize: make helper for add with saturation 2015-10-30 17:51:47 +01:00
Olivier Crête
7161795a44 liveadder: Make latency property be a uint in millisecs
This restores roughly the same behaviour as the old liveadder element.
Except that the latency now also includes the output-buffer-duration.

https://bugzilla.gnome.org/show_bug.cgi?id=757050
2015-10-28 18:52:24 -04:00
Wim Taymans
cd6c29e071 audioconvert: make the quantizer a reusable object
Turn the quantizer into a reusable object.
2015-10-28 11:36:18 +01:00
Wim Taymans
8fc2569328 audioconvert: make the channel mixer a separate reusable object
A first attempt at making the channel mixer a separate object.
2015-10-28 11:36:18 +01:00
Wim Taymans
8d4cd51e59 audioquantize: fix 8-pole noise shaping
Fix the 8-pole noise shaping error update. We were mixing errors from
different channels.
2015-10-28 11:36:18 +01:00
Sebastian Dröge
36b80edb72 decodebin: Send SEEK events directly to adaptive streaming demuxers
This makes sure that they will always get SEEK events, even if we're currently
in the middle of a group switch (i.e. switching to another
representation/bitrate/etc).

https://bugzilla.gnome.org/show_bug.cgi?id=606382
2015-10-27 15:50:45 +02:00
Guillaume Desmottes
7d6b6b0313 decodebin: fix event leak
As stated in GST_PAD_PROBE_HANDLED's documentation, we are
supposed to unref the event before returning.

Fixes an event leak in the validate.hls.playback.play_15s.hls_bibbop
validate scenario.

https://bugzilla.gnome.org/show_bug.cgi?id=754459
2015-10-25 11:18:29 +00:00
Sebastian Dröge
b4afaee8c0 audioconvert: Update disted orc files 2015-10-23 19:13:05 +03:00
Wim Taymans
2b626a5adf audioconvert: use pack/unpack functions
Rework the converter to use the pack/unpack functions
Because the unpack functions can only unpack to 1 format, add a separate
conversion step for doubles when the unpack function produces int.
Do conversion to S32 in the quantize function directly.
Tweak the conversion factor for doing float->int conversion slightly to
get the full range of negative samples, use clamp to make sure we don't
exceed our int range on the positive axis (see also #755301)
2015-10-23 16:58:17 +02:00
Sebastian Dröge
53f135cec7 playbin: Send upstream events directly to playsink
Send event directly to playsink instead of letting GstBin iterate
over all sink elements. The latter might send the event multiple times
in case the SEEK causes a reconfiguration of the pipeline, as can easily
happen with adaptive streaming demuxers.

What would then happen is that the iterator would be reset, we send the
event again, and on the second time it will fail in the majority of cases
because the pipeline is still being reconfigured
2015-10-23 12:02:28 +03:00
Thibault Saunier
ab6b536a66 videotestsrc: Force alpha downstream if foreground color contains alpha
Otherwise the foreground color won't be fully represented in the
outputted frames.

https://bugzilla.gnome.org/show_bug.cgi?id=755482
2015-10-22 11:12:23 +02:00
Matthew Waters
44871680f0 decodebin: track the exposable pads through connect_pad
The logic introduced by
[d50b713: decodebin: set the decode pad target before setting elements to PAUSED]
to expose pads would only ever be able to possibly expose one (the last) pad per element.

Make it so that any exposable pads are able to be exposed rather than just the
last pad returned by connect_element.

https://bugzilla.gnome.org/show_bug.cgi?id=742924
2015-10-20 10:48:05 +03:00
Matthew Waters
94d81fc713 decodebin: return the possibly new chain in analyze_new_pad
In the case of analyzing a demuxer chain, analyze_new_pad may create
a new GstDecodeChain.  This was not propagated to the calling function which as
of [d50b713f decodebin: set the decode pad target before setting elements to PAUSED]
is now required to be able to expose the correct pad.

https://bugzilla.gnome.org/show_bug.cgi?id=742924
2015-10-20 10:47:45 +03:00
Rajat Verma
68ec631db7 playsink: relink text_pad in case of reconfiguration
In case of reconfiguration, text_pad should be re-connected with
stream synchronizer sink pad. Otherwise we'll leave an unlinked pad around if
there always was a streamsynchronizer text pad.

https://bugzilla.gnome.org/show_bug.cgi?id=756804
2015-10-20 10:37:04 +03:00
Sebastian Dröge
4d6aa0f831 decodebin/playbin/playsink/subtitleoverlay: Post async-done on state change failures
https://bugzilla.gnome.org/show_bug.cgi?id=756611
2015-10-19 11:06:25 +03:00
Sebastian Dröge
87dbe54797 playsink: Immediately error out if state change fails
Otherwise we chain up to the parent class' change_state function and might
override the failure with SUCCESS.

https://bugzilla.gnome.org/show_bug.cgi?id=756611
2015-10-19 11:06:25 +03:00
Sebastian Dröge
92061cb19e playbin/uridecodebin: Always post async-done immediately if we're a live pipeline
Not only if the base class told us, but also if one of our own elements did.

https://bugzilla.gnome.org/show_bug.cgi?id=756611
2015-10-19 11:06:25 +03:00
Matthew Waters
d50b713f44 decodebin: set the decode pad target before setting elements to PAUSED
Otherwise caps and context queries will disappear into nothing and therefore
fail.  With autoplug-query now actually working, users (such as playbin) can
proxy these queries to the selected video sink and be able to select an
more appropriate configuration.

https://bugzilla.gnome.org/show_bug.cgi?id=731204
2015-10-19 11:55:04 +11:00
Vineeth TM
b424bc2e4f playsink: Fix volume element leak
In case sink implements a streamvolume interface, volume element is being got
from the sink. But this is transfer full. So the memory should be freed before
setting it to NULL. This was resulting in major memory leaks

https://bugzilla.gnome.org/show_bug.cgi?id=755867
2015-10-15 09:42:21 +03:00
Tim-Philipp Müller
45081ef6f1 liveadder: latency property is an uint64 in audiomixer 2015-10-12 09:42:37 +01:00
Olivier Crête
8cce2ccbf2 liveadder: Remove plugin, replace by compat subclass of audiomixer
New subclass with a similar behaviour as the old liveadder, but
a slightly different API as the latency is in nanoseconds, not
milliseconds. Also, the new liveadder has a effective latency that
is latency + output-buffer-duration. In practice, just setting a non-zero
latency with the new audiomixer gives you the right behavior in 99% of the
cases.
2015-10-11 11:04:38 +01:00
Vineeth TM
11ab8b7965 audioaggregator: Fix build error
Build error due to wrong argument type in debug message
aagg->priv->offset and next_offset are of type int64, but uint64
formatter is being used in logs. Changing all those to int64

https://bugzilla.gnome.org/show_bug.cgi?id=756065
2015-10-07 11:20:35 +01:00
Vineeth T M
a1d84edd16 videorate: remove unnecessary break statement
Trivial patch to remove unncessary break statement used after
goto statement.

https://bugzilla.gnome.org/show_bug.cgi?id=754054
2015-10-02 17:27:13 +03:00
Mathieu Duponchelle
0c0f803488 encodebin: Fix special case
Allows to run such a command line :

gst-launch-1.0 uridecodebin uri=file:///home/meh/Music/sthg.mp4 ! \
encodebin profile-string="audio/x-wav|1" ! filesink location=sthg.wav

Previously the code failed because wavenc is considered as a muxer.
We still want encodebin to audio/x-wav as an AudioEncodingProfile,
so this simple fix allows that.

Ability to mux raw streams in containers such as matroskamux
is a different issue.

https://bugzilla.gnome.org/show_bug.cgi?id=751470
2015-10-02 17:25:48 +03:00
Rajat Verma
267f4c2bad decodebin: free hidden groups at time of switching groups
hidden groups should be freed at time of switching groups to avoid memory use
from balloning up.

https://bugzilla.gnome.org/show_bug.cgi?id=755770
2015-10-02 17:02:21 +03:00
Jan Schmidt
3f7138a6e4 videotestsrc: Don't fixate framerate if downstream didn't provide one
intersection with a downstream that accepts any video/x-raw caps
with no further detail won't create a framerate field. If it's
not in the caps, don't fixate it, just set it to 30/1
2015-10-02 15:05:26 +10:00
Sebastian Dröge
52ea2667c6 audioaggregator: Select the initial offset based on the start segment position
instead of always using 0. Otherwise we might output a lot of silence in the
beginning instead of outputting from the relevant position.

https://bugzilla.gnome.org/show_bug.cgi?id=755623
2015-10-01 17:40:59 +02:00
Tim-Philipp Müller
fb30c04145 typefinding: minor clean-up
Remove unnecessary brackets from IS_MPEGTS_HEADER macro.
2015-10-01 12:49:59 +01:00
Pankaj Darak
eaf4ce01d3 typefinding: mpeg-ts detection improvement
Allow AFC to be 0 for null pid packets.

https://bugzilla.gnome.org/show_bug.cgi?id=726117
2015-10-01 12:32:33 +01:00
Tim-Philipp Müller
7fb9cd453b subparse: detect closing tags even if there's a space after the slash
</ i> should be handled like </i>

https://bugzilla.gnome.org/show_bug.cgi?id=755875
2015-09-30 18:17:13 +01:00
Tim-Philipp Müller
ddbae168b8 audiomixer: fix deadlock when G_DISABLE_ASSERT is not defined
This makes the audiomixer unit test time out in master.
Broke with 587e7c4
2015-09-26 10:21:41 +01:00
Sebastian Dröge
94342ca11a audioaggregator: Stop using deprecated gst_segment_to_position() 2015-09-26 00:17:55 +02:00
Sebastian Dröge
3eb52b293e audioaggregator: Only skip the remaining part of a GAP buffer
We might've queued up a GAP buffer that is only partially inside the current
output buffer (i.e. we received it too late!). In that case we should only
skip the part of the GAP buffer that is inside the current output buffer, not
also the remaining part. Otherwise we forward this pad too far into the future
and break synchronization.
2015-09-18 18:00:05 +02:00
Jan Schmidt
e88ecc367b Don't throw compiler warnings with G_DISABLE_ASSERT
Disable code that warns about unused variables when G_DISABLE_ASSERT
is defined, as it is in tarballs and pre-releases.
2015-09-18 00:29:51 +10:00
Vineeth T M
060f0c21f2 audiosink, multisocketsink: Fix error leak during failures
https://bugzilla.gnome.org/show_bug.cgi?id=755143
2015-09-17 11:59:35 +02:00
Sebastian Dröge
2a1e046dd9 uridecodebin: Use the correct caps name for MS Smooth Streaming manifests
Thanks to John Chang <r97922153@gmail.com> for reporting.

https://bugzilla.gnome.org/show_bug.cgi?id=755098
2015-09-16 19:54:43 +02:00
Sebastian Dröge
41fae5fa5d audioaggregator: Fix mixup of running times and segment positions
We have to queue buffers based on their running time, not based on
the segment position.

Also return running time from GstAggregator::get_next_time() instead of
a segment position, as required by the API.

Also only update the segment position after we pushed a buffer, otherwise
we're going to push down a segment event with the next position already.

https://bugzilla.gnome.org/show_bug.cgi?id=753196
2015-09-14 19:57:00 +02:00
Sebastian Dröge
c91e32bbf7 audioaggregator: Use stream time in the position query instead of segment position
https://bugzilla.gnome.org/show_bug.cgi?id=753196
2015-09-14 19:56:51 +02:00
Sebastian Dröge
35cb3b0c57 playback: Add POINTER_TO_ULONG() macro for consistency 2015-09-11 23:29:57 +02:00
Kouhei Sutou
ab64b00b48 playback: fix build error for 64bit Windows build by MinGW
Casting to gpointer from gulong generates the following warning with
64bit Windows target MinGW:

    gstplaybin2.c: In function 'pad_added_cb':
    gstplaybin2.c:3476:7: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
           (gpointer) group_id_probe_handler);
           ^
    cc1: all warnings being treated as errors

We should cast to guintptr from gulong before we cast to gpointer.

https://bugzilla.gnome.org/show_bug.cgi?id=754755
2015-09-11 23:28:07 +02:00
Wim Taymans
3d733ad09f videoscale: fix gamma-decode option
We need to use the enum to configure the option now.
2015-09-03 17:56:05 +02:00
Thiago Santos
76d26a60bd playsinkconvertbin: implement accept-caps handler
The default one will just go through the internal elements which might
just be identity when it is in passthrough which will lead to the query
being handled by the downstream sink, ignoring all that playsinkconvertbin
could actually handle and convert.

https://bugzilla.gnome.org/show_bug.cgi?id=754235
2015-08-28 09:44:10 -03:00
Eunhae Choi
b1f78b5d23 subparse: use g_clear_error instead of g_error_free
To avoid invalid pointer accees the err pointer should be set to NULL.
By using g_clear_error() it calls free and clear the pointer.

https://bugzilla.gnome.org/show_bug.cgi?id=753817
2015-08-19 16:21:25 +03:00
Sebastian Dröge
2727ca01f5 Revert "decodebin: Handle the preroll multi-queue size"
This reverts commit 5c8ef0ea05.
2015-08-18 18:47:22 +03:00
Sebastian Dröge
4fe4357188 Revert "decodebin: Store extra_buffer_required per group, not globally"
This reverts commit 1ea81114ea.
2015-08-18 18:47:21 +03:00
Sebastian Dröge
970bc16bf8 Revert "decodebin: If extra buffers are going to be required, we're still prerolling"
This reverts commit a3b24f0241.
2015-08-18 18:47:18 +03:00
Sebastian Dröge
a3b24f0241 decodebin: If extra buffers are going to be required, we're still prerolling 2015-08-18 15:19:03 +03:00
Sebastian Dröge
1ea81114ea decodebin: Store extra_buffer_required per group, not globally
It's only relevant for each group, and by storing it in the group
we have locking and everything else like for the other buffering-related
variables. Locking looks a bit fishy still, but it was like that for a long
time already so shouldn't be worse than before.
2015-08-18 15:19:03 +03:00
Myoungsun Lee
5c8ef0ea05 decodebin: Handle the preroll multi-queue size
Overview:
There are some of interleaved streams which has long-term location of audio data.
It mean the audio data is located far away more than multiqueue size.
In this case, because of multiqueue overrun, the pipeline is stopped.
To prevent hanging-like state, the decodebin needs to handle the queue size.

Caused:
The multiqueue size is not enough, the pipeline will stay being stalled status
and decodebin cannot complete to build decode chain.
In this issue file, decodebin did not receive no_more_pads signal or audio data yet.

Steps to Reproduce:
play the high-resolution(4K file) files or some streaming media(push mode).

Actual Results:
There is no audio or subtitle.
We can see only video or infinite loading.

Resolution:
Decodebin detect this problem, and add extra buffer size to multiqueue.
The multiqueue is larger than before, the next data can be pushed the downstream element.

Additional Information:
The max-preroll extra buffer size is set 8MB.
We can use total pre-roll buffer 10MB.
Only first overrun callback can handle multiqueue size.

https://bugzilla.gnome.org/show_bug.cgi?id=733235
2015-08-18 15:19:02 +03:00
Sebastian Dröge
8a736f6e98 typefindfunctions: Add typefinder for TTML+XML
Used in DASH among other things, as SMPTE Timed Text.
2015-08-18 12:56:33 +03:00
Edward Hervey
7fc856ff5c decodebin: Fix list iteration
We were using the wrong variable ...

CID #1316477
2015-08-16 12:53:23 +02:00
Edward Hervey
eaf9ca90c7 decodebin2: Handle flushing with multiple decode groups
When an upstream element wants to flush downstream, we need to take
all chains/groups into consideration.

To that effect, when a FLUSH_START event is seen, after having it
sent downstream we mark all those chains/groups as "drained" (as if
they had seen a EOS event on the endpads).

When a FLUSH_STOP event is received, we check if we need to switch groups.
This is done by checking if there are next groups. If so, we will switch
over to the latest next_group. The actual switch will be done when
that group is blocked.

https://bugzilla.gnome.org/show_bug.cgi?id=606382
2015-08-15 18:50:06 +02:00
Edward Hervey
2d3743e37d decodebin2: Forward event/queries for unlinked groups
When upstream events/queries reach sinkpads of unlinked groups (i.e.
no longer linked to the upstream demuxer), this patch attempts to find
the linked group and forward it upstream of that group.

This is done by adding upstream event/query probes on new group sinkpads
and then:
* Checking if the pad is linked or not (has a peer or not)
* If there is a peer, just let the event/query follow through normally
* If there is no peer, we find a pad to which to proxy it and return
  GST_PROBE_HANDLED if it succeeded (allowing the event/query to be properly
  returned to the initial called)

Note that this is definitely not thread-safe for the time being

https://bugzilla.gnome.org/show_bug.cgi?id=606382
2015-08-15 18:50:06 +02:00
Jan Schmidt
f188a023c7 typefind: Make the H.264 typefind a tiny bit more lenient.
When we see prefix NALs before a Subset SPS has been spotted,
it might just be because the stream was truncated at the
start, so don't count those as either 'bad' or 'good' packets.
2015-08-15 12:14:56 +10:00
Thiago Santos
052d1c7b8b playsinkconvertbin: remove accept-caps handling
Just let the internal element of the bin do it instead of forcing a
caps query to do it.
2015-08-14 05:48:31 -03:00
Thiago Santos
909f494a5a videorate: fixate the pixel-aspect-ratio
If the pixel-aspect-ratio is not fixed, try to get it as close
to 1/1 as possible

https://bugzilla.gnome.org/show_bug.cgi?id=748635
2015-08-13 14:11:37 -03:00
Joan Pau Beltran
5070d6367e videorate: add support for bayer formats
Since the videorate element just duplicates or drops frames
to achieve the desired framerate, it can accept video/x-bayer media
(in any format), which are not present in the current caps.
Just add "video/x-bayer(ANY);" to the caps of the static pad template
(fixing line style to pass the indent commit hook).

https://bugzilla.gnome.org/show_bug.cgi?id=753483
2015-08-10 17:21:03 -04:00
Vineeth TM
0a3fe31f26 decodebin: fix deadend_details string leak
deadend_details need not be returned when the pad is not a deadend.
Hence checking if res value is TRUE and clearing the string instead of
passing it on

https://bugzilla.gnome.org/show_bug.cgi?id=753088
2015-08-05 14:33:35 -03:00
Nicolas Dufresne
7db376d05e videotestsrc: Don't set DTS on buffer
DTS is for encoded data and have no meaning for raw. It better to not
set it, as it's confusing.

https://bugzilla.gnome.org/show_bug.cgi?id=752791
2015-08-04 18:00:35 -04:00
Olivier Crête
c2794d1ad0 audiointerleave: Avoid caps processing if not yet negotiated
https://bugzilla.gnome.org/show_bug.cgi?id=745768
2015-07-30 14:00:05 -04:00
Olivier Crête
f6507af946 audioaggregator: On timeout, resync pads with not enough data
https://bugzilla.gnome.org/show_bug.cgi?id=745768
2015-07-30 14:00:05 -04:00
Olivier Crête
08df711c0c aggregator: Queue "latency" buffers at each sink pad.
In the case where you have a source giving the GstAggregator smaller
buffers than it uses, when it reaches a timeout, it will consume the
first buffer, then try to read another buffer for the pad. If the
previous element is not fast enough, it may get the next buffer even
though it may be queued just before. To prevent that race, the easiest
solution is to move the queue inside the GstAggregatorPad itself. It
also means that there is no need for strange code cause by increasing
the min latency without increasing the max latency proportionally.

This also means queuing the synchronized events and possibly acting
on them on the src task.

https://bugzilla.gnome.org/show_bug.cgi?id=745768
2015-07-30 14:00:05 -04:00
Olivier Crête
9ab4b2e94e audioaggregator: Register function name
Otherwise, it sometimes segfaults with debugging enabled
2015-07-22 19:30:19 -04:00
Olivier Crête
ee1a50ef70 audioaggregator: Use 1.0 style buffer allocation 2015-07-22 19:30:12 -04:00
Nirbheek Chauhan
74d7944cbb audioaggregator: Sync pad values before aggregating
We need to sync the pad values before taking the aggregator and pad locks
otherwise the element will just deadlock if there's any property changes
scheduled using GstController since that involves taking the aggregator and pad
locks.

Also add a test for this.

https://bugzilla.gnome.org/show_bug.cgi?id=749574
2015-07-22 19:50:38 +01:00
Ville Skyttä
04e3fc3f33 typefind: Treat *.umx (Unreal Music Package) as audio/x-mod
https://bugzilla.gnome.org//show_bug.cgi?id=752436
2015-07-22 12:59:03 +01:00
Olivier Crête
a1cdede940 audioaggregator: Read output buffer duration with lock held 2015-07-21 21:55:25 -04:00
Tim-Philipp Müller
39576545b7 typefindfunctions: add DASH MPD typefinder
Moved from dashdemux plugin in -bad.
2015-07-16 21:26:30 +01:00
Tim-Philipp Müller
6020b0cf2b Update mailing list address from sourceforge to freedesktop 2015-07-16 17:17:16 +01:00
Wim Taymans
14083178b8 video: improve logging
Add logging categories for most video objects.
Remove some useless debug lines in video-info and videotestsrc.
Add a performance debug line in the video scaler.
2015-07-15 12:47:42 +02:00
Wim Taymans
2b2766494b socketsrc: add caps property
Add caps property that allows the src to easily negotiate a format.
2015-07-14 16:01:10 +02:00
Thiago Santos
9c2e08c54d decodebin: only try to expose complete groups
When switching to a new chain it might be that this new chain
is not yet ready to be exposed so check it before exposing.

Can happen with mpegts that might delay adding pads or pushing data
until it has found the PMT/PAT/PCR and that may take a while depending
on the stream.

It happened frequently with HLS:
http://vevoplaylist-live.hls.adaptive.level3.net/vevo/ch1/appleman.m3u8
2015-07-14 00:11:59 -03:00
Thiago Santos
1d1bebd769 decodebin: fix typo
Hided -> hid
2015-07-14 00:11:59 -03:00
Sebastian Dröge
f99a24f8b3 playsink: Require the streamvolume interface on the sink when using the sink's volume/mute properties
If the sink has properties named volume and mute, we have no idea about their
meaning. The streamvolume interface standardizes the meaning.

In the case of osxaudiosink for example, the current volume property has a
range of 0.0 to 1.0, but we need 0.0 to 10.0 or similar. Also osxaudiosink
has no mute property. As such, the volume element should be used here instead.

https://bugzilla.gnome.org/show_bug.cgi?id=752156
2015-07-10 11:55:23 +03:00
Thiago Santos
822b1d4511 typefind: also check moof to recognize video/quicktime
Helps recognizing fragmented files with the right type
2015-07-06 10:07:38 -03:00
Stefan Sauer
75cc08d451 docs: order and canonicalize the -sections.txt file
Have all sections in alphabetical order. Also make the macro order consistent.
This is a preparation for generating the file. Remove GET_CLASS macro for
some elements, since it is not used and the header is not installed.
2015-07-03 21:16:27 +02:00
Stefan Sauer
68c5adec38 videoscale: fix debug categories
Use a local category for the default category and fix the import for the
performance category.
2015-07-03 21:08:03 +02:00
danny song
49d0083456 playbin: remove unnecessary break
https://bugzilla.gnome.org/show_bug.cgi?id=751690
2015-06-29 19:55:34 -03:00
Sebastian Dröge
c5dbee33b0 audioresample: Also copy metas if their API has no tags attached to it
This is the default basetransform behaviour, being more strict than that
is not really useful.
2015-06-29 13:06:59 +02:00
Sebastian Dröge
010e35afa7 audioconvert: Also copy metas if their API has no tags attached to it
This is the default basetransform behaviour, being more strict than that
is not really useful.
2015-06-29 13:06:49 +02:00
Song Bing
d2e942ac02 streamsynchronizer: Unblock EOS wait when track switching.
sink_event () will blocked on EOS event. which will cause can't
send event when switch EOS track to non-EOS one.

https://bugzilla.gnome.org/show_bug.cgi?id=750761
2015-06-23 15:28:49 +02:00
Sebastian Dröge
bd508a343f streamsynchronizer: Don't wait for sparse streams when doing stream switches
Their stream-start event might come a bit later, like just before the first
buffer... and queues might run full before that happens.
2015-06-22 20:54:18 +02:00
Sebastian Dröge
152534611d streamsynchronizer: Add some more debug output 2015-06-22 20:29:52 +02:00
Sebastian Dröge
203b635d0c streamsynchronizer: Reset group start time when flushing
We reset the group start time to the running time of the start of the other
streams that are not flushed. This fixes seeking in gapless mode after the
first track has played.

https://bugzilla.gnome.org/show_bug.cgi?id=750013
2015-06-22 20:17:56 +02:00
Sebastian Dröge
ab79e50510 playbin: Reset suburi also when receiving an error message from the sub uridecodebin
http://bugzilla.gnome.org/show_bug.cgi?id=751118
2015-06-22 14:51:07 +02:00
Brijesh Singh
bcc9021071 playbin: free group->suburi on failure
If suburidecodebin is failed to negotiate (e.g file does not exist)
then free internal suburi variable so that 'current-suburi' property
returns correct status.

https://bugzilla.gnome.org/show_bug.cgi?id=751118
2015-06-22 14:48:42 +02:00
Tim-Philipp Müller
f5ad17871c typefinding: check for full UTF-8 BOM in MSS typefinder
https://bugzilla.gnome.org/show_bug.cgi?id=750802
2015-06-11 23:33:30 +01:00
Philippe Normand
d182e66bce typefindfunctions: UTF-8 MSS Manifest detection support
Check if the first bytes of data contain an UTF-8 BOM.

https://bugzilla.gnome.org/show_bug.cgi?id=750802
2015-06-11 19:46:02 +02:00
Sebastian Dröge
9c47e7d5e6 playbin: Check in autoplug_continue against the subtitle factory caps correctly
6a2f017bfa changed it to check the subtitle
factory caps if there is a text-sink but we fail to get its sinkpad. What
actually should be done here is to use the factory caps if there is no
text-sink at all.

https://bugzilla.gnome.org/show_bug.cgi?id=750785
2015-06-11 16:18:51 +02:00
Jan Schmidt
ca864ce46e playbin: Fix some warnings with clang around multiview enums
There is the GstVideoMultiviewMode enum and the
GstVideoMultiviewFramePacking, which is a subset of the
multiview modes, with the same values as the corresponding
types from the full enum. Do some casts and use the right
times to avoid implicitly using/passing GstVideoMultiviewFramePacking
when a GstVideoMultiviewMode is needed.
2015-06-11 23:01:48 +10:00
Jan Schmidt
383d8f02be playbin: Implement multiview frame-packing overrides
Add GstVideoMultiviewFramePacking enum, and the
video-multiview-mode and video-multiview-flags
properties on playbin.

Use a pad probe to replace the multiview information in
video caps sent out from uridecodebin.

This is a part implementation only - for full
correctness, it should also modify caps in caps events,
accept-caps and allocation queries.

https://bugzilla.gnome.org/show_bug.cgi?id=611157
2015-06-11 12:05:00 +10:00
Víctor Manuel Jáquez Leal
7b78a33dc6 playsink: fix the channel of color balance element
When traversing the color balance element channel list to find the one that
matches with the playsink proxy, the assignation was set to iterator of the
playsink proxy, not the balance element. Thus, the mapping to the values of
the balance element channel was wrong.

This patch fixes the assignation of the color balance element channel, so the
mapping to the channel of the color balance element is fixed.

https://bugzilla.gnome.org/show_bug.cgi?id=750691
2015-06-10 13:12:31 +02:00
Vineeth TM
50beddd474 playsink: cannot enable text flag while playing
when text playbin is not enabled in the beginning, then
video_srcpad_stream_synchronizer gets linked to videochain->sinkpad
and when we try to enable text bin during play, since it is already linked to videochain,
text chain does not get linked properly. Hence unlinking the same
before linking to text chain

https://bugzilla.gnome.org/show_bug.cgi?id=748908
2015-06-10 10:38:45 +02:00
Tim-Philipp Müller
f64ebd1d21 audiomixer: fix misleading documentation copied from adder 2015-06-09 14:37:36 +01:00
Sreerenj Balachandran
90bbb830d9 playback: Skip 'ANY' capsfeature while finding the count of common capsfeatures
https://bugzilla.gnome.org/show_bug.cgi?id=687182
2015-06-09 10:13:08 +02:00
Sreerenj Balachandran
16378a7de3 playback: Add gstplaybackutils.{h,c} to deploy the common subroutines
Bring some of the helper functions in gstplaybin2.c to new files
gstplaybackutils.{h,c} which can be utilized by other files
in gst/playback too.

https://bugzilla.gnome.org/show_bug.cgi?id=687182
2015-06-09 10:13:08 +02:00
Sebastian Dröge
baff8aa729 Release 1.5.1 2015-06-07 10:55:35 +02:00
Mathieu Duponchelle
2ad27e4c13 audioresample: copy metadata that only has the "audio" tag.
https://bugzilla.gnome.org/show_bug.cgi?id=750406
2015-06-04 19:16:40 +02:00
Mathieu Duponchelle
88484399c5 audioconvert: copy metadata that only has the "audio" tag.
https://bugzilla.gnome.org/show_bug.cgi?id=750406
2015-06-04 19:16:40 +02:00
Olivier Crête
0fbf2da1bb audiointerleave: Always have "channels" be the actual pad count
Don't force it anywhere

https://bugzilla.gnome.org/show_bug.cgi?id=750252
2015-06-01 19:43:20 -04:00
Olivier Crête
47d7b546c9 audiointerleave: Use the channel count from the set caps
This is the same number that was used to allocate the buffer
2015-06-01 19:42:49 -04:00
Thibault Saunier
dcfb8a83a5 encodebin: Add a way to enable/disabled a GstEncodingProfile
Summary:
So that the user can easily use the same encoding profile to render
with/without audio/video stream.

API:
  gst_encoding_profile_is_disabled
  gst_encoding_pofile_set_enabled

https://bugzilla.gnome.org/show_bug.cgi?id=749056
2015-06-01 10:22:31 +02:00
Jan Schmidt
db86cff74d videotestsrc: Document the solid-color pattern 2015-05-30 01:11:47 +10:00
Jan Schmidt
b14ede9332 playback: Document GST_PLAY_FLAG_SOFT_COLORBALANCE 2015-05-30 01:11:47 +10:00
George Kiagiadakis
c84f911cee videorate: update the caps framerate only in the GST_PAD_SINK transform_caps direction
When a stream has a variable framerate, videorate calculates it and
forces it on the output caps. However, the code in _transform_caps()
currently also does that if the transform is going in the opposite
direction (GST_PAD_SRC), so during a renegotiation it tries to force
upstream to use the calculated framerate and it fails.

https://bugzilla.gnome.org/show_bug.cgi?id=750032
2015-05-29 15:03:05 +02:00
Thiago Santos
12ac087807 playsink: use queue to avoid lock in audiotee audio branches
This part of pipeline is:

tee name=t ! visualizationbin ! streamsynchronizer name=s
t. ! s.

streamsynchronizer might block and it could starve the visualization
branch of the pipeline when it is enabled.

The visualization bin has queues internally but the other branch
that links the audiotee directly to the synchronizer is vulnerable
to block. Adding a queue between "t. ! s." fixes deadlocks.

https://bugzilla.gnome.org/show_bug.cgi?id=749676
2015-05-28 04:56:16 -03:00
Tim-Philipp Müller
39cbe25df7 gio: don't use soon-to-be-deprecated g_cancellable_reset()
From the API documentation: "Note that it is generally not
a good idea to reuse an existing cancellable for more
operations after it has been cancelled once, as this
function might tempt you to do. The recommended practice
is to drop the reference to a cancellable after cancelling
it, and let it die with the outstanding async operations.
You should create a fresh cancellable for further async
operations."

https://bugzilla.gnome.org/show_bug.cgi?id=739132
2015-05-19 18:52:41 +01:00
Tim-Philipp Müller
37aa31379f tcp: don't use soon-to-be-deprecated g_cancellable_reset()
From the API documentation: "Note that it is generally not
a good idea to reuse an existing cancellable for more
operations after it has been cancelled once, as this
function might tempt you to do. The recommended practice
is to drop the reference to a cancellable after cancelling
it, and let it die with the outstanding async operations.
You should create a fresh cancellable for further async
operations."

https://bugzilla.gnome.org/show_bug.cgi?id=739132
2015-05-19 18:50:41 +01:00
Stefan Sauer
af5c05caf1 Revert "doc: Workaround gtkdoc issue"
This reverts commit ff6c736fe0.

This is fixed by the gtk-doc 1.23 release.

<para> cannot contain <refsect2>:
http://www.docbook.org/tdg/en/html/para.html
http://www.docbook.org/tdg/en/html/refsect2.html
2015-05-18 20:16:32 +02:00
Stefan Sauer
9b9d1a6119 Revert "doc: Workaround gtkdoc issue"
This reverts commit df7ef3c35d.

This is fixed by the gtk-doc 1.23 release.
2015-05-18 20:01:49 +02:00
eunhae choi
1b755eb272 playbin: check the flags before set again
check the previous flags of playsink to avoid the reconfigure of playsink repeatedly

https://bugzilla.gnome.org/show_bug.cgi?id=749528
2015-05-18 10:01:04 +03:00
Nicolas Dufresne
891c7c6149 doc: Workaround gtkdoc issue
With gtkdoc 1.22, the XML generator fails when a itemizedlist is
followed by a refsect2. Workaround the issue by wrapping the
refsect2 into para.
2015-05-16 23:38:14 -04:00
Nicolas Dufresne
df7ef3c35d doc: Workaround gtkdoc issue
With gtkdoc 1.22, the XML generator fails when a itemizedlist is
followed by a refsect2. Workaround the issue by wrapping the refsect2
into para.
2015-05-16 23:33:55 -04:00
Stefan Sauer
015bd9285a playback: use the new gst_object api
Use gst_object_has_as_anchestor instead of the now deprecated _has_ancestor.
2015-05-15 14:49:47 +02:00
Tim-Philipp Müller
ec5c93f169 docs: update element example pipelines
- gst-launch -> gst-launch-1.0
- use autoaudiosink and audiovideosink more often
- review pipeline examples and descriptions
2015-05-10 11:38:19 +01:00
Vivia Nikolaidou
327efa9805 videoconvert: Expose some properties from the videoconverter API
Expose chroma resampler, alpha mode, alpha value, chroma mode, matrix mode,
gamma mode and primaries mode from the videoconverter API.

https://bugzilla.gnome.org/show_bug.cgi?id=749105
2015-05-08 15:17:06 +02:00
Vivia Nikolaidou
c9cfd0196f video-converter: Change some implicit string enums to real enums
GST_VIDEO_CONVERTER_OPT_ALPHA_MODE, GST_VIDEO_CONVERTER_OPT_CHROMA_MODE,
GST_VIDEO_CONVERTER_OPT_MATRIX_MODE, GST_VIDEO_CONVERTER_OPT_GAMMA_MODE and
GST_VIDEO_CONVERTER_OPT_PRIMARIES_MODE were G_TYPE_STRING with only a few valid
options. Changed those to real enums.

https://bugzilla.gnome.org/show_bug.cgi?id=749104
2015-05-08 15:13:54 +02:00
Sebastian Dröge
a73631a29d streamsynchronizer: Don't override segment.base from upstream with 0
Upstream might want to use it to properly map timestamps to running/stream
times, if we just override it with 0 synchronization will be just wrong.

For this we remove some old 0.10 code related to segment accumulation, and
remove some more code that is useless now, and accumulate the group start time
(aka segment.base offset) manually now.

https://bugzilla.gnome.org/show_bug.cgi?id=635701
2015-05-05 15:35:46 +02:00
Luis de Bethencourt
69f66aff9e Rename property enums from ARG_ to PROP_
Property enum items should be named PROP_ for consistency and readability.
2015-04-27 11:27:00 +01:00
Matthieu Bouron
9dfe40d740 videoconvert: Keep colorimetry and chroma-site fields if passthrough
https://bugzilla.gnome.org/show_bug.cgi?id=748141
2015-04-27 11:20:36 +02:00
Tim-Philipp Müller
4984828fcf Remove obsolete Android build cruft
This is not needed any longer.
2015-04-26 18:59:32 +01:00
Tim-Philipp Müller
c680e324bc Remove obsolete Android build cruft
This is not needed any longer.
2015-04-26 18:42:34 +01:00
Tim-Philipp Müller
1638859627 typefinding: don't read more data than needed in MSS typefinder 2015-04-26 14:45:44 +01:00
Tim-Philipp Müller
9ef16721ed typefinding: detect MSS manifests without using g_convert()
Embedded systems often have limited charset conversion
functionality, so don't rely on g_convert() (i.e. iconv)
for UTF-16 to UTF-8 conversions, we can easily enough do
that ourselves by converting to native endianness and
then using GLib's helper functions.
2015-04-26 14:41:30 +01:00
Luis de Bethencourt
df08f5eabe remove unused enum items PROP_LAST
This were probably added to the enums due to cargo cult programming and are
unused. Removing them.
2015-04-24 17:11:01 +01:00
Bernhard Miller
1c01b50ada audioconvert: fix mixed usage of gint and gint32 in int matrix
This is a fixup for b2db18cda2
audioconvert: avoid float calculations when mixing integer-formatted channels

The int matrix was using gint and gint32 synonymously, which can theoretically
cause problems if gint and gint32 are actually different types.

https://bugzilla.gnome.org/show_bug.cgi?id=747005
2015-04-15 12:18:03 +02:00
Tim-Philipp Müller
dc4e517dc6 gio: fix gvfs plugin dependencies
Try harder to look for gvfs backend changes in the right
place, to make sure the plugin gets reloaded when backends
are removed or installed. We watch the gvfs mounts directory
because the files there contain absolute paths to the
backend executables, and those may not be in the usual gio
path.

https://bugzilla.gnome.org/show_bug.cgi?id=747841
2015-04-14 16:00:42 +01:00
Tim-Philipp Müller
d4f9ea8499 app, videorate: fix CFLAGS and LIBADD order
Make sure local headers are included before installed -base.
2015-04-11 00:03:29 +01:00
Sebastian Dröge
3570100b66 decodebin: Also log the pointer value of sticky events in debug output
Makes it easier to follow them in the debug logs.
2015-04-08 20:49:39 -07:00
Tim-Philipp Müller
6db2ee56b6 tcpserversink: don't error out if clients send us something, just ignore it
We don't expect clients to send us any data, but if they do, just
ignore it. Web browsers might send us an HTTP request for example,
but some will still be happy if we just send them data without
a proper HTTP response.

There was a bug in the reading code path. We only have a small
read buffer and would provoke an EWOULDBLOCK trying to read
because we don't bail out of the loop early enough.

https://bugzilla.gnome.org/show_bug.cgi?id=743834
2015-04-04 21:38:40 +01:00
Tim-Philipp Müller
609d021f96 videorate: downgrade left-over ERROR debug message 2015-04-04 00:49:23 +01:00
Tim-Philipp Müller
413fc30235 videorate: fix a couple of memory leaks
tests: videorate: fix leak in unit test
2015-04-04 00:49:21 +01:00
Vincent Penquerc'h
77dc09c3a9 decodebin2: fix deadlock on chain shutdown
When shutting down the chain, we can get a deadlock when removing
a pad, if that chain was being busy streaming but blocked (eg, while
waiting for a queue to have free space).

https://bugzilla.gnome.org/show_bug.cgi?id=746480
2015-04-03 15:42:49 +01:00
Thibault Saunier
ae86dec9ca videorate: Detect framerate if not forced to variable downstream
In case upstream does not provide videorate with framerate information,
it will detect the current framerate from the buffer it received,
but if downstream forces the use of variable framerate (most probably
through the use of a caps filter with framerate = 0 / 1), videorate will
respect that.

And add some unit tests

https://bugzilla.gnome.org/show_bug.cgi?id=734424
2015-04-02 17:13:24 -04:00
Thibault Saunier
1cda538e00 videorate: Do not loop forever pushing first buffer when variable framerate
In the case the framerate is variable (represented by framerate=0/1),
we currently end up loop pushing the first buffer and then recompute
diff1 and diff2 without updating the videorate->next_ts at all
leading to infinitely looping pushing that first buffer.

In the case of variable framerate, we should just compute the next_ts
as previous_pts + previous_duration.

https://bugzilla.gnome.org/show_bug.cgi?id=734424
2015-04-02 17:13:24 -04:00
Olivier Crête
5d78c5cca6 audiomixer: Allow downstream caps with a non-default channel-mask
Instead of failing, take the downstream channel mask if the channel
count is 1.
2015-04-01 20:32:41 -04:00
Bernhard Miller
b2db18cda2 audioconvert: avoid float calculations when mixing integer-formatted channels
The patch calculates a second channel mixing matrix from the current one. The
matrix contains the original values * (2^10) as integers. This matrix is used
when integer-formatted channels are mixed.

On a ARM Cortex-A8, single core, 800MHz this improves performance in a
testcase from 29s to 9s for downmixing 6 channels to stereo.

https://bugzilla.gnome.org/show_bug.cgi?id=747005
2015-04-01 07:31:37 -07:00
Luis de Bethencourt
985ed4847f playbin: avoid possible deference of null pointer
For safety, check the pointer playbin->curr_group is valid before
reading parameters of the structure.

CID #1291624
2015-03-30 10:50:48 +01:00
Thiago Santos
ceb26dd93d decodebin: improve debug message by printing the object
Print the pad object that EOS'd too early
2015-03-27 09:21:59 -03:00
Thiago Santos
d54d51d0d2 playbin: ignore new pads if it is shutting down
If a new pad is added after playbin has been put to READY/NULL it
should ignore new pads as it is shutting down.

This can happen when the pipeline fails to preroll (is still in READY)
and the user gives up on waiting or an error that doesn't reach
the demuxer occurs (on some event handling) and it will continue to
work and exposing pads while playbin has been put to NULL.

Without this check an input-selector is created and set to PAUSED
state, preventing playbin from properly shutting down in case it
has data blocked inside it.
2015-03-25 08:32:33 -03:00
Nicolas Dufresne
9695222b0f videorate: Don't leak the pools
gst_query_set_nth_alloction_pool() is transfer none on the pool, so we must
unref the pool when done.
2015-03-24 15:23:34 -04:00
Luis de Bethencourt
1011a50766 audioaggregator: check sink caps are valid 2015-03-24 16:18:22 +00:00
Luis de Bethencourt
8199405dd7 Revert "audioaggregator: check sink caps are valid"
This reverts commit 6d4d0d1cdf.

Never put code with side effects into an assertion, it can be compiled out
2015-03-24 16:17:00 +00:00
Luis de Bethencourt
a7cfb6240f audioaggregator: check sink caps are valid
CID #1291622
2015-03-24 15:53:17 +00:00
Ilya Konstantinov
3dc3aa4e3b audioconvert: Eliminate unsigned quantizers
audio_convert_convert unpacks to default format (signed) before calling
quantize, and the unsigned variants were equivalent to signed anyway,
so we just get rid of them.
2015-03-24 16:52:07 +01:00
Ilya Konstantinov
7b398701cf audioconvert: Avoid int division in quantization
Since range size is always 2^n, we can simply use modulo (implemented
with a bitmask).

The previous implementation used 64-bit integer division, which is
done in software on ARMv7. Although the divisor was constant, the
division could not be transformed into "multiplication by magic number"
since the dividend was 64-bit.

The now-unused and not-so-fast gst_fast_random_(u)int32_range functions
were removed.

Also, implementing bug fixes:

1) ADD_DITHER_TPDF_HF_I no longer discards bias.

2) We change TPDF's noise range to be the same as RPDF's. Previously,
RPDF's noise ranged:
  { bias - dither, bias + dither }
while TPDF's noise ranged:
  { bias/2 - dither/2, bias/2 + dither/2 - 1 } +
  { bias/2 - dither/2, bias/2 + dither/2 - 1 } =
  { bias - dither, bias + dither - 2 }
Now, both range:
  { bias - dither, bias + dither - 1 }

https://bugzilla.gnome.org/show_bug.cgi?id=746661
2015-03-24 16:52:07 +01:00
Duncan Palmer
bf3e35a598 decodebin2: Set multiqueue sizes before use-buffering.
This fixes a race where the use-buffering property on a multiqueue was
set before the queue depth was changed from it's high preroll limits to
lower playback limits. This resulted in buffering messages being emitted
by the multiqueue in the short window between use-buffering being
set and the queue depth being reset.

https://bugzilla.gnome.org/show_bug.cgi?id=744308
2015-03-24 08:17:47 -03:00
Olivier Crête
edde3c326e audiointerleave: Set src caps in aggregate
This prevents races between the setcaps of the sink pads

https://bugzilla.gnome.org/show_bug.cgi?id=740236
2015-03-16 16:44:03 -04:00
Olivier Crête
fb8339de40 audiointerleave: Add interleave element based on audioaggregator
https://bugzilla.gnome.org/show_bug.cgi?id=740236
2015-03-16 16:44:03 -04:00
Olivier Crête
15369ba016 audioaggregator: Print a message when a buffer is late
https://bugzilla.gnome.org/show_bug.cgi?id=740236
2015-03-16 16:44:03 -04:00
Olivier Crête
acf7745188 audioaggregator: Don't re-send the caps if they did not change
https://bugzilla.gnome.org/show_bug.cgi?id=740236
2015-03-16 16:41:45 -04:00
Olivier Crête
1eef58c3ce audioaggregator: Split base class from audiomixer
Also:
-  Don't modify size on early buffer
   The size is the size of the buffer, not of remaining part.
- Use the input caps when manipulating the input buffer
   Also store in in the sink pad
- Reply to the position query in bytes too
- Put GAP flag on output if all inputs are GAP data
- Only try to clip buffer if the incoming segment is in time or samples
- Use incoming segment with incoming timestamp
   Handle non-time segments and NONE timestamps
- Don't reset the position when pushing out new caps
- Make a number of member variables private
- Correctly handle case where no pad has a buffer
  If none of the pads have buffers that can be handled, don't claim to be EOS.
- Ensure proper locking
- Only support time segments

https://bugzilla.gnome.org/show_bug.cgi?id=740236
2015-03-16 16:41:45 -04:00
Olivier Crête
66807c14fd audiomixer: Release pad object lock before dropping buffer
Otherwise, the locking order is violated and deadlocks happen.

https://bugzilla.gnome.org/show_bug.cgi?id=745768
2015-03-16 14:31:50 -04:00
Olivier Crête
3b2bc85ec6 audiomixer: Only ignore pads with no buffers on timeout
When the timeout is reached, only ignore pads with no buffers, iterate
over the other pads until all buffers have been read. This is important
in the cases where the input buffers are smaller than the output buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=745768
2015-03-16 14:31:50 -04:00
Olivier Crête
3f59bc95b8 audiomixer: Only advance by the buffer size when a buffer is late
https://bugzilla.gnome.org/show_bug.cgi?id=745768
2015-03-16 14:31:50 -04:00
Thiago Santos
b0b0ae1f24 streamsynchronizer: fix deadlock condition
The variables could have changed when the lock was released
to push a gap event. Streamsynchronizer needs to check them
again before going to sleep.

Bonus: fix a comment typo
2015-03-14 18:14:07 +00:00
Ramiro Polla
33b9535d97 playsink: remove redundant else statements 2015-03-14 14:21:32 +00:00
Ramiro Polla
b636fe29f3 playbin: don't escape percent sign in documentation code sample 2015-03-14 14:20:17 +00:00
William Manley
8328eab2de socketsrc: Add support for GstNetControlMessageMeta
multisocketsink now understands the new GstNetControlMessageMeta to allow
sending control messages (ancillary data) with data when writing to Unix
domain sockets.

Thanks to glib's `GSocketControlMessage` abstraction the code introduced
in this commit is entirely portable and doesn't introduce and additional
dependencies or conditionally compiled code, even if it is unlikely to be
of much use on non-UNIX systems.
2015-03-14 13:23:28 +01:00
William Manley
e63e023e30 multisocketsink: Add support for GstNetControlMessageMeta
multisocketsink now understands the new GstNetControlMessageMeta to allow
sending control messages (ancillary data) with data when writing to Unix
domain sockets.

A later commit will introduce a new socketsrc element which will similarly
understand `GstNetControlMessageMeta`.  This, when used with a
`GSocketControlMessage` of type `GUnixFDMessage` will allow GStreamer to
send and receive file-descriptions in ancillary data, the first step to
using memfds to implement zero-copy video IPC.

Thanks to glib's `GSocketControlMessage` abstraction the code introduced
in this commit is entirely portable and doesn't introduce and additional
dependencies or conditionally compiled code, even if it is unlikely to be
of much use on non-UNIX systems.
2015-03-14 13:23:20 +01:00
William Manley
a297b0545f socketsrc: Add connection-closed-by-peer signal
This provides notification that the socket in use was closed by the peer
and gives an opportunity to replace it with a new one which is not
closed, allowing reading from many sockets in order.

I use this in pulsevideo to implement reconnection logic to handle the
pulsevideo service dieing, such that is can be restarted without
disrupting downstream.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=739546
2015-03-13 20:05:00 +01:00
William Manley
a19ac4b85c socketsrc: Tidy up usage of g_object_unref/g_clear_object and locking
This is clearer, and should make future changes safer.  No functional
change intended.

See https://bugzilla.gnome.org/show_bug.cgi?id=739546
2015-03-13 20:05:00 +01:00
William Manley
0c054aa00d socketsrc: Refactor to simplify
* Don't bother polling, just do a blocking read, the `GCancellable` will
  take care of unlocking.  This should also be faster on MS Windows where
  the GIO documentation for `g_socket_get_available_bytes` states: "Note
  that on Windows, this function is rather inefficient in the UDP case".

* Implement `GstPushSrc.fill` rather than `GstPushSrc.create`.  This means
  that we will be using the downstream allocator which may be more
  efficient.  It also means that socketsrc is likely to respect its
  "blocksize" property (assuming that there is enough data available).

See https://bugzilla.gnome.org/show_bug.cgi?id=739546
2015-03-13 20:05:00 +01:00
William Manley
7c10499ecd tcp: Add element socketsrc
`socketsrc` can be considered a source counterpart to `multisocketsink`.
It can be considered a generalization of `tcpclientsrc` and
`tcpserversrc`:  it contains all the logic required to communicate over
the socket but none of the logic for creating the sockets/establishing
the connection in the first place, allowing the user to accomplish this
externally in whatever manner they wish making it applicable to other
types of sockets besides TCP.

This commit essentially copies the implementation directly from
tcpserversrc.  Later patches will tidy the implementation up and
re-implement `tcpclientsrc` and `tcpserversrc` in terms of `socketsrc`.

See https://bugzilla.gnome.org/show_bug.cgi?id=739546
2015-03-13 20:05:00 +01:00
William Manley
b8232a7467 multisocketsink: Map GstMemorys individually when sending
If a buffer is made up of non-contiguous `GstMemory`s `gst_buffer_map`
has to copy all the data into a new `GstMemory` which is contiguous.  By
mapping all the `GstMemory`s individually and then using scatter-gather
IO we avoid this situation.

This is a preparatory step for adding support to multisocketsink for
sending file descriptors, where a GstBuffer may be made up of several
`GstMemory`s, some of which are backed by a memfd or file, but I think this
patch is valid and useful on its own.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=746150
2015-03-13 16:20:50 +01:00
Sebastian Dröge
25d8f76ecd audiomixer: Fix discont detection and buffer alignment code
Actually accumulate the sample counter to check the accumulated error
between actual timestamps and expected ones instead of just resetting
the error back to 0 with every new buffer.

Also don't reset discont_time whenever we don't resync. The whole point of
discont_time is to remember when we first detected a discont until we actually
act on it a bit later if the discont stayed around for discont_wait time.

https://bugzilla.gnome.org/show_bug.cgi?id=746032
2015-03-12 17:14:33 +00:00
Jan Schmidt
3d60fb654b docs: Add new video functions and objects. Cleanup a little.
Add GstVideoChroma, GstVideoDither, GstVideoScaler and friends to the docs.

Remove and clean up a few obsolete/deleted refs and typos
2015-03-13 01:08:25 +11:00
Sebastian Dröge
8093e3ba94 playbin: Disconnect signals and invalidate group if it fails to activate
Otherwise playbin might move to the group directly after EOS of the next
group, and then error out again.
2015-03-12 12:18:30 +00:00
Nirbheek Chauhan
4e221b7a65 audiomixer: Add locking to fill_buffer and fix mix_buffer
The audiomixer pad struct fields may be changed from other threads
2015-03-12 09:53:28 +00:00
Nirbheek Chauhan
8227310d22 audiomixer: Mark a discont when we receive a new segment event
This allows us to handle new segment events correctly; either by dropping
buffers or inserting silence; for example if the offset is changed on an srcpad
connected to audiomixer.
2015-03-12 09:52:15 +00:00
Song Bing
7ce97c723c streamsynchronizer: Remove unnecessary ERROR message.
Remove unnecessary ERROR message.
Push GAP will fail as flushing. Needn't ERROR message.

https://bugzilla.gnome.org/show_bug.cgi?id=736655
2015-03-11 15:48:42 +00:00
Wim Taymans
9bbfc3c848 videotestsrc: add all colors mode 2015-03-10 12:27:03 +01:00
Tim-Philipp Müller
c53ba4beeb Fix double semicolons 2015-03-10 09:27:08 +00:00
Olivier Crete
124b6ee03c videorate: Accept any capsfeatures 2015-03-09 21:39:02 -04:00
Sebastian Dröge
40f4daffd1 volume: Explicitly cast integers to doubles and then back to integers after multiplication
gcc 4.9.1 on ARM seems to have a bug that causes it to cast the float to an
integer first, resulting in a 0 scale factor for volume < 1.0.

As a side effect this change here will also improve accuracy of the result a
bit because we go via doubles instead of floats.

https://gcc.gnu.org/bugzilla/show_bug.cgi?id=65325
https://bugzilla.gnome.org/show_bug.cgi?id=745667
2015-03-05 14:22:12 +01:00
Sebastian Dröge
38cf87aaea Revert "audiomixer: Latency is twice the output buffer duration, not only once"
This reverts commit d387cf67df.

The analysis was wrong: The first 20ms of latency are introduced by the source
already and put into the latency query, making it only necessary to cover the
additional 20ms of audiomixer inside audiomixer.
2015-03-04 13:16:03 +01:00
Sebastian Dröge
fc917fb8cf audiomixer: Latency is twice the output buffer duration, not only once
Let's assume a source that outputs outputs 20ms buffers, and audiomixer having
a 20ms output buffer duration. However timestamps don't align perfectly, the
source buffers are offsetted by 5ms.

For our ASCII art picture, each letter is 5ms, each pipe is the start of a
20ms buffer. So what happens is the following:

0   20  40  60
OOOOOOOOOOOOOOOO
|   |   |   |

  5   25  45  65
  IIIIIIIIIIIIIIII
  |   |   |   |

This means that the second output buffer (20 to 40ms) only gets its last 5ms
at time 45ms (the timestamp of the next buffer is the time when the buffer
arrives). But if we only have a latency of 20ms, we would wait until 40ms
to generate the output buffer and miss the last 5ms of the input buffer.
2015-03-03 20:06:48 +01:00
Arun Raghavan
dfdbc1c379 adder: Drop custom latency querying logic
The default latency query handler now implements the same logic already.
2015-02-27 00:52:05 +05:30
Edward Hervey
7813315a4c playback: Fix broken GList modification
When we modify a GList (via g_list_delete_link), always reassign the
new head to the original GList. Otherwise we end up with
filtered_errors being corrupt (the head might have been the element
removed)
2015-02-26 12:08:49 +01:00
Vincent Penquerc'h
561ddabd97 decodebin: fix deadlock when resetting buffering
This function is static, and only ever called with the expose lock
taken. It thus has no reason to take this lock itself.

This was introduced by one of my locking fixes from 741355.

https://bugzilla.gnome.org/show_bug.cgi?id=741355
2015-02-24 16:07:26 +00:00
Vincent Penquerc'h
a2ca6459a6 playbin: forward template and ring buffer settings to existing decodebins
https://bugzilla.gnome.org/show_bug.cgi?id=744844
2015-02-24 10:02:19 +00:00
Luis de Bethencourt
8703d93bbf decodebin: move null check
Check if dbin->decode_chain is NULL before running drain_and_switch_chains()
because if it is, we shouldn't run that function or it will segfault.

CID #1271074
2015-02-23 17:24:56 +00:00
Sebastian Dröge
1dcd1a7479 decodebin: Only consider non-parser factories for generating the post-parser capsfilter caps
Otherwise if there are multiple parsers we would most likely break negotiation
of the stream-format/alignment wanted by the decoders as parsers generally
support all possible stream-formats and alignments.
2015-02-20 12:35:19 +02:00
Vincent Penquerc'h
a2ee84fa80 decodebin: fix deadlock between downward state change and pad addition
If caps on a newly added pad are NULL, analyze_new_pad will try to
acquire the chain lock to add a probe to the pad so the chain can
be built later. This comes from the streaming thread, in response
to headers or other buffers causing this pad to be added, so the
stream lock is taken.

Meanwhile, another thread might be destroying the chain from a
downward state change. This will cause the chain to be freed with
the chain lock taken, and some elements are set to NULL here, which
can include the parser. This causes pad deactivation, which tries
to take the element's pad's stream lock, deadlocking.

Fix this by keeping track of which elements need setting to NULL,
and only do this after the chain lock is released. Only the chain
manipulation needs to be locked, not the elements' state changes.

https://bugzilla.gnome.org/show_bug.cgi?id=741355
2015-02-19 13:38:35 +00:00
Vincent Penquerc'h
a848ac7abe decodebin: guard against the decode chain going while a pad is added
https://bugzilla.gnome.org/show_bug.cgi?id=741355
2015-02-19 13:38:35 +00:00
Vincent Penquerc'h
9036dc8594 decodebin: possible fix for deadlock when spamming "next song"
There was a deadlock between a thread changing decodebin/demuxer
state from PAUSED to READY, and another thread pushing data
when starting.

From the stack trace at
https://bug741355.bugzilla-attachments.gnome.org/attachment.cgi?id=292471,
I deduce the following is happening, though I did not reproduce the
problem so I'm not sure this patch fixes it.

The streaming thread (thread 2 in that stack trace) takes the demuxer's
sink pad's stream lock in gst_ogg_demux_perform_seek_pull and will
activate a new chain. This ends up causing the expose lock being taken
in _pad_added_cb in decodebin.

Meanwhile, a state changed is triggered on thread 1, which takes the
expose lock in decodebin in gst_decode_bin_change_state, then frees
the previous chain, which ends up calling gst_pad_stop_task on the
demuxer's task, which in turn takes the demuxer's sink pad's stream
lock, deadlocking as both threads are now waiting for each other.

https://bugzilla.gnome.org/show_bug.cgi?id=741355
2015-02-19 13:38:29 +00:00
Sebastian Dröge
2813e08210 uridecodebin: Reset the default query return value when the iterator has to resync 2015-02-19 01:30:05 +02:00
Sebastian Dröge
5da04ca3c7 uridecodebin: Let the latency query fail if one of the source queries fails 2015-02-19 01:22:26 +02:00
Olivier Crête
0487e1548d uridecodebin: Pass object, not GValue to debug print 2015-02-17 18:39:03 -05:00
Song Bing
fb9ca25f7f streamsynchronizer: Use the same waiting function for EOS and stream switches
Also improve the waiting condition for stream switches, which was assuming
before that the condition variable will only stop waiting once when it is
signaled. But the documentation says that there might be spurious wakeups.

https://bugzilla.gnome.org/show_bug.cgi?id=736655
2015-02-16 14:34:35 +02:00
Song Bing
2614f80309 streamsynchronizer: Send GAP events from the pads' streaming threads
Change the GAP events that are currently sent from the chain function of
the current pad to all other EOS pads. They should instead be sent from
their own streaming threads.

https://bugzilla.gnome.org/show_bug.cgi?id=736655
2015-02-16 14:12:28 +02:00
Song Bing
9f81931716 streamsynchronizer: Send GAP event to finish preroll when change state from PLAYING to PAUSED
Wait in the event function when EOS is received until all pads are EOS
and then forward the EOS event from each pads own event function.

Also send a new GAP event for EOS pads from the event function whenever
going from PLAYING->PAUSED by shortly waking up the GCond. This is needed
to allow sinks to pre-roll again, as they did not receive EOS yet because
we blocked that, but also will never get data again.

https://bugzilla.gnome.org/show_bug.cgi?id=736655
2015-02-16 14:09:43 +02:00
Tim-Philipp Müller
5fc7d39090 audiomixer: use new gst_aggregator_pad_drop_buffer() 2015-02-13 16:25:52 +00:00
Tim-Philipp Müller
195e54e06a audiomixer: calculate stream_time used to sync pad values correctly
Use pad (input) segment to calculate the stream time from the
input timestamp, not the aggregator (output) segment.
2015-02-12 11:41:10 +00:00
Stefan Sauer
c51bf98af4 playbin: improve debug log
Log the human readable pad_link_return desc as well.
2015-02-11 22:16:53 -08:00
Sebastian Dröge
8547594727 Improve and fix LATENCY query handling
This now follows the design docs everywhere, especially the maximum latency
handling.

https://bugzilla.gnome.org/show_bug.cgi?id=744106
2015-02-11 17:53:49 +02:00
Sebastian Dröge
3c9ae895b0 Improve and fix LATENCY query handling
This now follows the design docs everywhere, especially the maximum latency
handling.

https://bugzilla.gnome.org/show_bug.cgi?id=744106
2015-02-11 14:16:21 +01:00
Tim-Philipp Müller
68515c4439 audiomixer: remove now-unused base_time field in object structure 2015-02-06 10:47:20 +00:00
Sebastian Dröge
6d6c693254 audiomixer: Remove weird and wrong segment handling
There's no reason why audiomixer should override the segment
base of upstream with whatever value it got from a SEEK event,
or even worse... with 0 if there was no SEEK event yet. This
broke synchronization if upstream provided a segment base other
than 0, e.g. when using pad offsets.
Also that this code did things conditional on the element's state
should've been a big warning already that something is just wrong.
If this breaks anything else now, let's fix it properly :)

Also don't do fancy segment position trickery when receiving a
segment event. It's just not correct.
2015-02-05 16:02:54 +01:00
Luis de Bethencourt
f85212ed4b videoscale: fix memory leak
In gst_video_scale_fixate_caps () it can goto done without freeing the memory
of the tmp GstStructure. This makes it go out of scope and leak.

CID #1265766
2015-02-04 12:09:45 +00:00
Thibault Saunier
b1eef4f436 aggregator: Make the PAD_LOCK private
Instead of using the GST_OBJECT_LOCK we should have
a dedicated mutex for the pad as it is also associated
with the mutex on the EVENT_MUTEX on which we wait
in the _chain function of the pad.

The GstAggregatorPad.segment is still protected with the
GST_OBJECT_LOCK.

Remove the gst_aggregator_pad_peak_unlocked method as it does not make
sense anymore with a private lock.

https://bugzilla.gnome.org/show_bug.cgi?id=742684
2015-01-29 10:24:18 +01:00
Thibault Saunier
198b16c563 aggregator: Hide GstAggregatorPad buffer and EOS fileds
And add a getter for the EOS.

The user should always use the various getters to access
those fields

https://bugzilla.gnome.org/show_bug.cgi?id=742684
2015-01-29 10:24:18 +01:00
Olivier Crête
9afd2b3339 audiomixer: Clear GstAudioInfo the the caps
When clearing the caps, also clear the matching GstAudioInfo

https://bugzilla.gnome.org/show_bug.cgi?id=742684
2015-01-29 10:24:18 +01:00
Olivier Crête
33f412d6db audiomixer: Don't reset caps on flush
A flush event doesn't invalidate the previous caps event.

https://bugzilla.gnome.org/show_bug.cgi?id=742684
2015-01-29 10:24:18 +01:00
Olivier Crête
9071b8487c aggregator: Replace event lock with pad's object lock
Reduce the number of locks simplify code, what is protects
is exposed, but the lock was not.

Also means adding an _unlocked version of gst_aggregator_pad_steal_buffer().

https://bugzilla.gnome.org/show_bug.cgi?id=742684
2015-01-29 10:24:18 +01:00
Wim Taymans
47bd6a138c videoscale: don't do dithering 2015-01-28 17:30:53 +01:00
Luis de Bethencourt
783204824d orc: update orc files 2015-01-27 13:39:14 +00:00
Wim Taymans
3db8879f25 videoscale: disable chroma and matrix operations
Ignore chroma subsampling and color matrix transformations like the
old videoscale used to do. This is to make the performance like it was
before.

See https://bugzilla.gnome.org/show_bug.cgi?id=741987
2015-01-27 10:52:29 +01:00
Sebastian Dröge
2bd4ea6e8e Constify some static arrays everywhere 2015-01-21 09:49:47 +01:00
Vincent Penquerc'h
661588b150 dcodebin2: fix lock/unlock mismatch on multiqueue overrun 2015-01-20 15:09:13 +00:00
Jan Alexander Steffens (heftig)
a636c39638 audioresample: Try to prevent endless looping
Speex may decide not to consume any samples because it can't write any. I've
seen a hang during draining caused by the resample loop never terminating.
In that case, resampling happened as normal until olen was 0 but ilen was
still 1. _process_native then reduced ichunk to 0, so ilen never decreased
below 1 and the loop never terminated.

Instead of reverting 684cf44 ({audioresample: don't skip input samples),
break only if all output samples have been produced and speex refuses
to consume any more input samples.

https://bugzilla.gnome.org/show_bug.cgi?id=732908
2015-01-19 19:36:13 +01:00
Sebastian Dröge
63afbce6be videorate: Add $(GST_PLUGINS_BASE_CFLAGS) to be able to find gst/video/video.h 2015-01-19 11:17:18 +01:00
Nicolas Dufresne
e60158c98f videorate: Implement allocation query
The videorate element keeps 1 buffer internally. This buffer need
to be requested during allocation query otherwise the pipeline may
stall.

https://bugzilla.gnome.org/show_bug.cgi?id=738302
2015-01-18 14:58:36 -05:00
Nicolas Dufresne
2e264103e1 Revert "videorate: Implement allocation query"
This reverts commit 3c04db4a30.
2015-01-18 14:17:07 -05:00
Nicolas Dufresne
3c04db4a30 videorate: Implement allocation query
VideRate keeps 1 buffer in order to duplicate base on closest buffer
relative to targeted time. This extra buffer need to be request
otherwise the pipeline may stall when fixed size buffer pool is used.

https://bugzilla.gnome.org/show_bug.cgi?id=738302
2015-01-18 11:02:00 -05:00
Sebastian Dröge
2228d9f22b decodebin: Fix compilation 2015-01-17 14:51:48 +01:00
Branislav Katreniak
d16df7f70d decodebin: do call set_queue_size in no_more_pads_cb
Consider pipeline: gst-launch-1.0 playbin uri=http://example.com/a.ogg
Consider 128kbit audio stream.

As soon as uridecodebin detects the bitrate, it configures its input
queue2 max-size to 32000 bytes.
The 2MB buffer in multiqueue is nearly 2 orders of magnitude bigger.
This non-deterministically drives queue2 buffer anywhere from
100% to 0% until multiqueue is filled.

This patch sets multiqueue size to 5 buffers early in no_more_pads_cb.

Partly reverts commit db771185ed.

https://bugzilla.gnome.org/show_bug.cgi?id=740689
2015-01-16 20:58:40 +01:00
Vincent Penquerc'h
6ab711f3f1 decodebin: free old groups when switching groups
Old groups are freed with one switch's delay when switching groups.
They're freed in a scratch thread to avoid delaying the switch.
2015-01-16 15:55:10 +00:00
Thiago Santos
a5ed7afb4c decodebin: disable pad link checks as it has already been done
Decodebin has already added the element to the bin and should only
select caps compatible pads. It should disable the pad link checks
to avoid doing those again.

https://bugzilla.gnome.org/show_bug.cgi?id=742885
2015-01-14 10:33:52 -03:00
Tim-Philipp Müller
d7880f217e audiomixer: update for aggregator start/stop vfunc change 2014-12-30 18:01:34 +00:00
Tim-Philipp Müller
5cf0b8c445 audiomixer: fix output-block-size property description 2014-12-30 15:32:46 +00:00
Nirbheek Chauhan
ecc709be31 audiomixer: Document the pad properties 2014-12-27 11:02:36 +00:00
Sebastian Dröge
8abfdd127f videotestsrc: Report our latency properly in live mode
While we have no latency at all in theory, any other live source has the
duration of one buffer as minimum latency. Do the same in videotestsrc.

https://bugzilla.gnome.org/show_bug.cgi?id=741879
2014-12-24 12:59:37 +01:00
Sebastian Dröge
631d356845 audiotestsrc: Report our latency properly in live mode
While we have no latency at all in theory, any other live source has the
duration of one buffer as minimum latency. Do the same in audiotestsrc.

https://bugzilla.gnome.org/show_bug.cgi?id=741879
2014-12-24 12:59:37 +01:00
Sebastian Dröge
cd256acf03 audiomixer: If getting a timeout before having caps, just advance our position
This can happen if this is a live pipeline and no source produced any buffer
and sent no caps until the an output buffer should've been produced according
to the latency.
2014-12-23 12:24:48 +01:00
Sebastian Dröge
eefea80dae audiomixer: Make sure to release the current buffer in reset()
If we didn't output the last one in aggregate because we were shutting down
earlier we might otherwise leak it.
2014-12-23 12:15:50 +01:00
Sebastian Dröge
8465c0915e audiomixer: Change blocksize property to output-buffer-duration in time format
This makes the interface of audiomixer independent of the actual caps.
2014-12-23 11:45:50 +01:00
Sebastian Dröge
20a79bda49 audiomixer: Use the src query implementation of aggregator as the default case 2014-12-22 22:12:02 +01:00
Stefan Sauer
b2fef1f9d2 audiomixer: fix build flag order
Have the libraries/inlcudes from plugins-bad first to avoid picking up the installed version.
Fixes the build when the local api changed.
2014-12-21 07:47:25 -05:00
Sebastian Dröge
bf3896b2bd audiomixer: Track discont-time per pad instead of globally
We do discont handling per pad, not per element!
2014-12-19 14:40:33 +01:00
Sebastian Dröge
bc418c7a85 audiomixer: We're only EOS if all our pads are actually EOS
Having a buffer or not on the pad is irrelevant.
2014-12-18 23:33:58 +01:00
Sebastian Dröge
eff64c7ddc audiomixer: The pad's size is always supposed to be the whole buffer size
And the offset the offset into that buffer. Changing the size will
cause all kinds of assumptions to fail and cause crashes.
2014-12-18 22:42:14 +01:00
Sebastian Dröge
06f6d3c65c aggregator: Add function to allow subclasses to set their own latency
For audiomixer this is one blocksize, for videoaggregator this should
be the duration of one output frame.
2014-12-17 19:51:32 +01:00
Sebastian Dröge
46f713c598 audiomixer: Make sure to not have pads being behind the current offset
We would break sync between the different streams then.
2014-12-17 19:37:22 +01:00
Sebastian Dröge
d508b39952 aggregator: Add a timeout parameter to ::aggregate()
When this is TRUE, we really have to produce output. This happens
in live mixing mode when we have to output something for the current
time, no matter if we have enough input or not.
2014-12-17 18:41:41 +01:00
Sebastian Dröge
cf90f534f1 audiomixer: Implement get_next_time() 2014-12-16 17:37:12 +01:00
Sebastian Dröge
6521870077 Revert "decodebin: Only emit the drain signal for the main decode chain, not any subchains"
This reverts commit a391dfe17f.

It breaks gapless playback: https://bugzilla.gnome.org/show_bug.cgi?id=740045
2014-12-15 09:46:13 +01:00
Matej Knopp
4713694082 audiorate: Fill gap events
https://bugzilla.gnome.org/show_bug.cgi?id=741281
2014-12-14 12:09:12 +01:00
Thibault Saunier
b9cbfcdeb4 playbin: Do not mix up stream type when getting stream combiner element
We were always returning the video stream combiner whatever stream type
combiner was wanted.
2014-12-11 13:47:58 +01:00
Thiago Santos
7e801a5f26 playbin2: always unref the combiner sinkpad when removing the srcpad
Create a function to do the pad cleanup of the GstSourceCombine struct
and use it to not forget to also cleanup the sink pad and fix a memory
leak.

https://bugzilla.gnome.org/show_bug.cgi?id=741198
2014-12-10 13:36:37 -03:00
Edward Hervey
6a2f017bfa playbin: Only check sinks which are in >= GST_STATE_READY
Otherwise we endup with bogus caps intersection (from the pad template
caps and not from what the actual hardware/device supports)

https://bugzilla.gnome.org/show_bug.cgi?id=738131
2014-12-05 07:58:44 +01:00
Chad
e397b03f35 audiorate: Use gst_util_uint64_scale_int_round()
Using gst_util_uint64_scale_int() causes slight drift
which accumulates over time.

https://bugzilla.gnome.org/show_bug.cgi?id=741045
2014-12-02 16:07:05 -05:00
Wim Taymans
991a81bd5d videoconvert: add dither-bits option
Fix the dither option.
Add a new option to set the quantizer
2014-12-02 15:23:00 +01:00
Sebastian Dröge
90eb93c2ef Don't compare booleans for equality to TRUE and FALSE
TRUE is 1, but every other non-zero value is also considered true. Comparing
for equality with TRUE would only consider 1 but not the others.
2014-12-01 09:51:12 +01:00
Thibault Saunier
72c05d1cbb encodebin: Add a way to disable caps renegotiation for output stream format
In some cases, the user might want the stream outputted by encodebin to
be in the exact same format during all the stream. We should let the
user specify when this is the case. This commit add some API in the
GstEncodingProfile to determine whether the format can be renegotiated
after the encoding started or not.

API:
    gst_encoding_profile_set_allow_dynamic_output
    gst_encoding_profile_get_allow_dynamic_output

https://bugzilla.gnome.org/show_bug.cgi?id=740214
2014-11-28 16:56:32 +01:00
Thibault Saunier
d5a171cae9 audiomixer: Do not try to resize a buffer to a negative size on EOS 2014-11-27 19:10:58 +01:00
Wim Taymans
9efbba8c1c videoscale: use old property name
Unbreak ABI by changing to the old property name again.

https://bugzilla.gnome.org/show_bug.cgi?id=740798
2014-11-27 09:47:41 +01:00
Thibault Saunier
35f6259b24 decodebin: Analyze source pad before setting to PAUSED for 'simple demuxers'
Before we were setting them to PAUSED and (much) later connecting to
their source pad caps notify signal.

There was a race where that demuxer was pushing a caps and later a buffer
on its source pad when we were not even connected to its source pad caps notify
signal leading to decodebin missing the information and not keeping on
building the pipeline on CAPS event thus the demuxer was posting an ERROR
(not linked) message on the bus. This need to be done for 'simple
demuxers' because those have one ALWAYS source pad, not like usual demuxers
that have several dynamic source pads.

A "simple demuxer" is a demuxer that has one and only one ALWAYS source
pad.

https://bugzilla.gnome.org/show_bug.cgi?id=740693
2014-11-26 19:38:48 +01:00
Mathieu Duponchelle
68edf0ebd6 decodebin2: Take STREAM_LOCK before sending sticky events.
There was a race where:

1) we would put the element to PAUSED
2) It would get data sent to it from upstream
3) It would thus send caps
3) caps_notify_cb would continue autoplugging
4) caps would flow downstream, the last pad would get exposed
5) we were still not done sending the sticky events

Taking the stream lock on the new element's sinkpad and only
releasing it when sticky events have all been sent prevents
the caps from reaching the source pad of the element before
we're all set.

https://bugzilla.gnome.org/show_bug.cgi?id=740694
2014-11-26 19:38:48 +01:00
Tim-Philipp Müller
76199fddb6 typefindfunctions: detect mp4 common file format variant
Used e.g. by UltraViolet.
2014-11-26 16:54:39 +00:00
Wim Taymans
43f44d41dc videoscale: add property to do scaling after gamma-decode 2014-11-25 11:54:51 +01:00
Wim Taymans
724b83c5f5 videoscale: add more scaling filters
Adjust the filter parameters so that they use the same number of taps
and method as the old ones.
Add some new filters
2014-11-25 11:28:42 +01:00
Andrei Sarakeev
e5c6f59140 playsink: Reset mute property of the sink to playsink's value when setting up the audio chain
Otherwise the following can happen:
1. set mute=true
2. play media1 (Ok)
3. play media without audio (audiochain removed)
4. play media2 (audiochain created, mute=*false*)

https://bugzilla.gnome.org/show_bug.cgi?id=740675
2014-11-25 10:23:50 +01:00
Tim-Philipp Müller
ef23ac5f52 typefind: improve 'audible' audio typefinder a little
Don't return NEARLY_CERTAIN just based on 4 bytes.
Also change media type to audio/x-audible.

https://bugzilla.gnome.org/show_bug.cgi?id=715050
2014-11-25 09:04:37 +00:00
Jonathan Matthew
ddda5866c5 typefindfunctions: add audio/audible typefinder
https://bugzilla.gnome.org/show_bug.cgi?id=715050
2014-11-25 00:55:50 +00:00
Sebastian Dröge
cefc518494 videorate: Operate in a zero-latency mode if drop-only is set to TRUE
There's no reason why we would have to wait for the next buffer to decide
whether to output the current one or not. We just have to check if the
current one is earlier than our expected next time, which is the previous
frame timestamp plus the expected frame duration.

https://bugzilla.gnome.org/show_bug.cgi?id=740018
2014-11-19 15:10:48 +01:00
Tim-Philipp Müller
9c5d53c7e3 docs: fix some gtk-doc warnings
Deprecated entities found in documentation for xyz:Long_description
.
2014-11-15 23:13:42 +00:00
Wim Taymans
835422b2ea videoscale: port to new API 2014-11-10 11:45:52 +01:00
Peter G. Baum
16c8856b42 typefind: recognize Apple Core Audio Format
(CAF) Specification 1.0

https://bugzilla.gnome.org/show_bug.cgi?id=739840
2014-11-09 14:42:40 +00:00
William Manley
ffb43c0591 tcpserversink: Don't leak a GSocket and a GInetSocketAddress
when accepting a connection.

Discovered by `make check-valgrind` with the new `socketintegrationtest`.

https://bugzilla.gnome.org/show_bug.cgi?id=739544
2014-11-07 10:15:43 +01:00
Andreas Frisch
bae96c85ee subtitleoverlay: return available factory CAPS instead of ANY on CAPS query
https://bugzilla.gnome.org/show_bug.cgi?id=739536
2014-11-03 08:20:13 +01:00
Vincent Penquerc'h
5d1376cefa typefind: remove unneeded test
We've already bailed out if we have less than 5 bytes.

Coverity 1226441
2014-10-30 11:42:02 +00:00
Wim Taymans
71efeaaa6f video-convert: swap src and dest
It is more natural and consistent with other uses.
2014-10-29 16:26:10 +01:00
Sebastian Dröge
8b8b8ae2e8 Revert "decodebin: fix the autoplugging of parser elements"
This reverts commit 2b0d392741.

This breaks cases where an actual second parser is required after the parser,
e.g. to do timestamp corrections.

See https://bugzilla.gnome.org/show_bug.cgi?id=738416
2014-10-26 11:04:47 +01:00
Sebastian Dröge
2da56de19f Revert "decodebin: Fix locking"
This reverts commit aa94d5dc9a.
2014-10-26 11:04:38 +01:00
Sebastian Dröge
aa94d5dc9a decodebin: Fix locking
The chain mutex needs to be locked when looking at chain->elements. Move code
around a bit to require only one lock() and unlock().
2014-10-21 13:32:19 +02:00
Sreerenj Balachandran
2b0d392741 decodebin: fix the autoplugging of parser elements
If there are two parser elements available for the same media format,
then decodebin is autoplugging an extra capsfilter and parser irrespective
of caps and rank. So restrict the decodebin from autoplugging multiple parser
elements back to back in adjacent positions with in a single DecodeChain
for the same media format.

https://bugzilla.gnome.org/show_bug.cgi?id=738416
2014-10-21 13:32:19 +02:00
Vineeth T M
c2224b8059 videotestsrc: assertion error
timestamp_offset is being declared as an int64 variable,
for which the min
value of G_MININT64 is -9223372036854775808
Changing the minimum and maximum limit for the offset variable.

https://bugzilla.gnome.org/show_bug.cgi?id=738568
2014-10-21 11:28:11 +02:00
Sreerenj Balachandran
a24db77217 decodebin: optimize the code a bit by avoiding unnecessary string comparisons
https://bugzilla.gnome.org/show_bug.cgi?id=738416
2014-10-21 11:05:53 +02:00
Sreerenj Balachandran
f60da86ae2 decodebin: Fix typo in comment
https://bugzilla.gnome.org/show_bug.cgi?id=738416
2014-10-21 11:05:53 +02:00
Vineeth T M
d38e242a72 audiomixer: critical error for blocksize, timeout min/max values
Audiomixer blocksize, cant be 0, hence adjusting the minimum value to 1
timeout value of aggregator is defined with MAX of MAXINT64,
but it cannot cross G_MAXLONG * GST_SECOND - 1
Hence changed the max value of the same

https://bugzilla.gnome.org/show_bug.cgi?id=738845
2014-10-21 10:58:48 +02:00
Stefan Sauer
7c247ab166 typefind: use gslice for typefine data
Also use our free function in the failure case.
2014-10-17 12:47:50 +02:00
Tim-Philipp Müller
d0aa9f9a34 encodebin: fix some leaks in error code path
Fixes test_encodebin_sink_pads_nopreset_static
running under valgrind.
2014-10-14 16:54:44 +01:00
Heinrich Fink
4497b73806 playsink: Use correct property enum value for video-filter property installation 2014-10-10 12:14:57 +03:00
Matthew Waters
57c8272c75 aggregator: add latency query handling 2014-10-09 23:52:11 +11:00
Luis de Bethencourt
f638f3b32b videoscale: remove FIXME about NV21 support
NV21 is already supported so removing FIXME about adding support for it.
2014-10-08 16:50:52 +01:00
Wim Taymans
042b25542c videotestsrc: add gradient pattern
Makes a gradient between background and foreground color.
2014-10-08 11:36:33 +02:00
Peter G. Baum
0b4abc267e audioresample: remove unused variables
https://bugzilla.gnome.org/show_bug.cgi?id=738026
2014-10-07 14:59:10 +03:00
Danny Song
bb6ea450e8 typefindfunctions: Remove leftover #define from 0.10
https://bugzilla.gnome.org/show_bug.cgi?id=738018
2014-10-07 14:54:01 +03:00
Andrei Sarakeev
a391dfe17f decodebin: Only emit the drain signal for the main decode chain, not any subchains
https://bugzilla.gnome.org/show_bug.cgi?id=738064
2014-10-07 14:48:54 +03:00
Thibault Saunier
22da31f42a audiomixer: Handle seek event in READY state 2014-10-06 18:57:28 +02:00
Thibault Saunier
c158e019c1 audiomixer: Set the sinkpad segments basetime after seeking
Otherwise stream offset and running time comparison will not be
correct, leading to segfaults after seeks
2014-10-06 18:57:28 +02:00
Thibault Saunier
183f4b3227 audiomixer: Port to GstAggregator
https://bugzilla.gnome.org/show_bug.cgi?id=737183

Co-Authored by: Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
2014-10-06 18:57:28 +02:00
Sebastian Dröge
72eb84a900 decodebin: Free factories array when delaying autoplugging due to non-final caps 2014-10-06 10:15:13 +03:00
Aurélien Zanelli
796fd16550 decodebin: unref decode pad after usage
https://bugzilla.gnome.org/show_bug.cgi?id=737757
2014-10-06 09:53:39 +03:00
Andres Gomez
09872442f8 uridecodebin: Removed setting "iradio-mode" property in the source element
The "iradio-mode" property used to have a default FALSE value in HTTP
source elements but now it should default to TRUE or just do not exist
as a property so it is not really needed to set it any more in
uridecodebin.

Apart from that this code could've never worked as uridecodebin looks for a
string-typed iradio-mode property, but it's a boolean in all sources.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725383
2014-10-02 09:50:48 +03:00
Wim Taymans
2977ef5281 videotestsrc: storel is better then copyl
It is better to use storel to splat the variable into the destination.
ORC doesn't know when a variable is last written to so it can't yet optimize
away the copy operation.
2014-09-26 16:44:09 +02:00
Luis de Bethencourt
a4e9e1fe1b videoscale: avoid recalculating values
Avoid recalculating values used multiple times as base of index. Plus some style
fixes.

https://bugzilla.gnome.org/show_bug.cgi?id=737400
2014-09-26 15:31:55 +01:00
Ravi Kiran K N
f16cf75194 videoscale: support lanczos method for NV formats
Support lanczos scaling method for NV12 and NV21 formats.
Scale the 'Y' plane and scale 'NV' plane.
Implementation for submethods - int16, int32, float and double

https://bugzilla.gnome.org/show_bug.cgi?id=737400
2014-09-26 13:05:18 +01:00
Wim Taymans
98c42dc5e4 video: convertor -> converter 2014-09-24 16:19:30 +02:00
Wim Taymans
b2fd20c416 video: move videoconvert code to video library
Move the conversion code used in videoconvert to the video library
and expose a simple but generic API to do arbitrary conversion. It can
currently do colorspace conversion but the plan is to add videoscale to
it as well.

See https://bugzilla.gnome.org/show_bug.cgi?id=732415
2014-09-24 15:59:39 +02:00
Wim Taymans
0c40b83ed4 video-color: add gst_video_color_matrix_get_Kr_Kb()
Move the function to get the color matrix coefficients from
videoconvert to the video library.
2014-09-24 15:59:39 +02:00
Sebastian Dröge
5adff2e65a videoscale Use stride instead of width in more places 2014-09-23 23:12:19 +03:00
Sanjay NM
db296c5924 videoscale: Use width instead of stride in buffer offset calculation
https://bugzilla.gnome.org/show_bug.cgi?id=736944
2014-09-23 23:08:54 +03:00
Sanjay NM
6babe786f7 videoscale: Added NV support for 4Tap resize
https://bugzilla.gnome.org/show_bug.cgi?id=736845
2014-09-18 13:31:08 +03:00
Andrei Sarakeev
2133a98eb1 playbin: Don't leak input-selector sinkpads
https://bugzilla.gnome.org/show_bug.cgi?id=736861
2014-09-18 12:52:02 +03:00
Ognyan Tonchev
d9e67777e0 streamsplitter: do not leak events when flushing them
https://bugzilla.gnome.org/show_bug.cgi?id=736796
2014-09-18 12:40:22 +03:00
Ravi Kiran K N
cf98138c7e typefind: correct the condition for irap flag
https://bugzilla.gnome.org/show_bug.cgi?id=736779
2014-09-17 09:41:36 +03:00
Sebastian Dröge
52e97f59ba playsink: Add audio/videoconvert in front of the audio/video-filters
audioresample and videoscale is something the application will have to do if
required, but we can at least help here by adding the
audioconvert/videoconvert elements.

https://bugzilla.gnome.org/show_bug.cgi?id=735748
2014-09-16 21:42:46 +03:00
Thiago Santos
3657929e1f decodebin: protect buffering message handling
Use the object lock to avoid concurrent processing which leads
to small disasters (assertions or crashes)
2014-09-11 17:16:18 -03:00
George Kiagiadakis
a2122f04ec playbin: filter out buffering messages when switching uri
When switching URI from about-to-finish, playbin starts decoding the new
URI and the queue2 inside uridecodebin starts emitting buffering messages
immediately. However, the queue(s) inside playsink still have buffers to
play and the pipeline doesn't need to pause for buffering, so we should
not send those buffering messages up to the application, otherwise there
is an audible glitch caused by pausing the pipeline for a very short time.

https://bugzilla.gnome.org/show_bug.cgi?id=727255
2014-09-05 12:44:27 -03:00
Kipp Cannon
684cf44ee3 audioresample: don't skip input samples
when downsampling, the output buffer can be filled before all the input
samples are consumed.  this is correct:  when downsampling, several input
samples are needed for each output sample, so when only a small number of
input samples are available the number of output samples produced can be 0.

the resampler, however, was discarding those extra input samples instead of
clocking them into its filter history for the next iteration.  this patch
fixes this by removing the check that the output buffer is full.  the code
now always loops until all input samples are consumed, and relies on the
calling code to have provided a suitably sized location for the output.
note that there are already other checks in place in the calling code to
ensure that this is the case.

https://bugzilla.gnome.org/show_bug.cgi?id=732908
2014-09-05 11:17:43 +03:00
Vineeth T M
302f123c62 videorate: GstStructure refcount critical message
s3 is not being initialized when run in a loop
and the same was being freed, which resulted in the crash

https://bugzilla.gnome.org/show_bug.cgi?id=735952
2014-09-03 12:56:31 +03:00
Sebastian Dröge
5cbefaa9a2 decodebin: Also include the raw caps in the error message, not just the human readable description 2014-09-02 15:37:38 +03:00
Sebastian Dröge
56899b596e decodebin: Include codec description for missing plugins in the error message
If we had plugins and an error occurred we only include the error message
caused by this, otherwise we will include the codec description as generated
from the caps.

This allows to detect which exact codec was missing instead of getting a
generic "no suitable decoders found" error message.
2014-09-02 13:00:48 +03:00
Tim-Philipp Müller
db857e5a97 encoding: remove assignment that's no longer needed
CID 1231980
2014-08-29 18:21:13 +01:00
Sebastian Dröge
2434af3d31 playsinkconvertbin: setcaps() always returns TRUE and the return value is unused
Change it to a void return value. The caps are forwarded afterwards via
gst_pad_event_default() and not inside this function.

CID 1226477
2014-08-28 17:13:05 +03:00
Sebastian Dröge
a5cf0a4572 decodebin: Include information from the error messages of tried but failed elements in the missing plugin errors 2014-08-25 21:01:16 +03:00
Sebastian Dröge
22a138b716 decodebin: Initialize local variables for every retry 2014-08-25 21:01:16 +03:00
Sebastian Dröge
21e9f84486 decodebin: Remove error case that resulted in two error messages
We already send one in gst_decode_bin_expose() for this case. Only
if we're unable to typefind the caps another error message is needed.
2014-08-25 21:01:16 +03:00
Tim-Philipp Müller
f14494f425 typefinding: tighten checks for 'freeform mp3' a little
Freeform mp3s typically have bitrates higher than the
otherwise max allowed rate. Prevents misdetection of
some truetype font files as mp3.

https://bugzilla.gnome.org/show_bug.cgi?id=732923
2014-08-25 11:18:21 +01:00
Thiago Santos
98ed3ddc8f playsinkconvertbin: only intersect with the filter at the end
Otherwise we might change some capsfeatures from ANY to the specific
value from the filter and do not filter those out in case the
sink doesn't support them

https://bugzilla.gnome.org/show_bug.cgi?id=734822
2014-08-15 18:24:36 -03:00
Thiago Santos
14d79a3a47 decodebin: handle group switching for deadend group
Gracefully handle switching groups that all pads are deadend.

This can happen when quickly switching programs on mpegts as the
output is unaligned it can happen that not enough data was accumulated at
parsers to generate any buffers, causing the stream to receive EOS before
any data can be decoded.

To handle this scenario, the _expose function now also gets if there is
any next group to be exposed along with the list of endpads. If there are
no endpads and there is another group to expose it will switch to this next
group and then retry exposing the streams.

Also, the requirement to only switch from the chain that has the endpad had
to be modified to care for when the drainpad is NULL

https://bugzilla.gnome.org/show_bug.cgi?id=733169
2014-08-13 18:51:37 +03:00
Thiago Santos
9c09c8ae17 decodebin: consider all deadend pads as drained
Otherwise when switching out a group with a deadend pad it will block
as it would be waiting for EOS on a deadend that already got one

https://bugzilla.gnome.org/show_bug.cgi?id=733169
2014-08-13 18:51:37 +03:00
Sebastian Dröge
d280bba126 playsinkconvertbin: Make sure to intersect raw caps with our converter caps
Otherwise we end up allowing video/x-raw with arbitrary caps features that are
not handled by our converters.

https://bugzilla.gnome.org/show_bug.cgi?id=734683
2014-08-13 14:28:05 +03:00
Sanjay NM
8cab1ab5fc videoscale: Add NV21 support
https://bugzilla.gnome.org/show_bug.cgi?id=734650
2014-08-12 14:31:48 +03:00
Thiago Santos
c9904fb639 encodebin: delay missing encoder error as passthrough is still possible
Set up a fakesink with a pad probe to replace the missing encoder to detect
if encoding was really required and only error out in this case. Otherwise
just let passthrough branch work.

This delays the error posting from the set_state function to when buffers
are really flowing. Unit test updated accordingly

https://bugzilla.gnome.org/show_bug.cgi?id=650652
2014-08-11 10:30:58 -03:00
Sebastian Dröge
59fb749ef6 decodebin: Remove buffering special casing for adaptive streaming demuxers
They output smaller buffers now and we should be able to handle the buffering
limits like in every other situation now.
2014-08-11 10:57:43 +02:00
Sebastian Dröge
a0a9fd004b playbin: Keep a reference to the playsink sinkpads
Otherwise playsink might get shut down without us noticing
that our pad references are gone now.

Probably fixes https://bugzilla.gnome.org/show_bug.cgi?id=733165
2014-08-01 15:00:46 +02:00
Mohammed Sameer
b34e0ba91c streamsynchronizer: don't unset DISCONT flag
Unsetting DISCONT flag means we need to copy the buffer. This copy operation
mandates that all GstMemory should be copy-able which is not always the case

https://bugzilla.gnome.org/show_bug.cgi?id=727409
2014-08-01 14:23:07 +02:00
Thiago Santos
cf50b45ff6 decodebin: add missing 'time' word to debug message
It prints the buffers, bytes and time limits, but 'time' was missing
from the string.
2014-07-29 15:55:27 -03:00
Sebastian Dröge
362e9a547b playbin: Pass through NO_PREROLL state change returns
Fixes playback of live pipelines.
2014-07-28 16:57:00 +02:00
Sebastian Dröge
f3f55e1758 uridecodebin: Pass through NO_PREROLL state change returns
Fixes playback of live pipelines.
2014-07-28 16:57:00 +02:00
Tim-Philipp Müller
1ed192abb0 playbin: fix 'attempt to unlock mutex that was not locked' in error code path
Fixes playbin unit test with latest GLib.
2014-07-26 14:52:01 +01:00
Sebastian Dröge
f173fa15b1 decodebin: Don't unref caps for which we don't own a reference... get one first
https://bugzilla.gnome.org/show_bug.cgi?id=733615
2014-07-23 19:51:36 +02:00
Sebastian Dröge
73646bd04f playbin: Go asynchronously from READY to PAUSED
We now add all our elements to uridecodebin *after*
GstBin::change_state(READY->PAUSED), so we need to post async-start
and async-done messages ourselves if we want to work async.

https://bugzilla.gnome.org/show_bug.cgi?id=733495
2014-07-23 12:46:48 +02:00
Sebastian Dröge
5c038192e2 uridecodebin: Go asynchronously from READY to PAUSED
We now add all our elements to uridecodebin *after*
GstBin::change_state(READY->PAUSED), so we need to post async-start
and async-done messages ourselves if we want to work async.

https://bugzilla.gnome.org/show_bug.cgi?id=733495
2014-07-23 12:46:48 +02:00
Sebastian Dröge
c051b378d7 uridecodebin: Create new sources after chaining up to the parent class
Otherwise we start the new sources already before the parent class
got ready to start.
2014-07-21 09:35:36 +02:00
Sebastian Dröge
5bf3c92462 playbin: Create new sources after chaining up to the parent class
Otherwise we start the new sources already before the parent class
got ready to start.
2014-07-21 09:35:36 +02:00
Sebastian Dröge
b15a47aa19 decodebin: Link Parser/Converter directly and already connect to pad-added and other signals before setting elements to PAUSED
otherwise we're going to
a) start Parser/Converter before they are linked to their capsfilter,
   breaking their negotiation of a proper stream format
b) start demuxers without having connected to their pad-added signals. We
   miss pads and in the worst case don't link any pads at all
2014-07-21 09:35:36 +02:00
Sebastian Dröge
57999c28fd decodebin: Send sticky events to the new element after setting it to PAUSED
... and if this fails for whatever reason we skip the element and instead
try with the next element. This allows us to handle elements that fail
when setting caps on them by just skipping to the next alternative element.
2014-07-21 09:35:36 +02:00